UC320 SIP Default Caller ID

Hello,
I post on this forum because I'm configuring cisco UC320 (Latest Firmware 2.3.2)
I use the UC320 only with SIP account.
For the most phones, I configure a inbound calling with specific inbound number and I check the case "use <number> like call ID".
When my phone that has a direct number call, I see the correct number on the called phone.
But when an internal phone that has not a inbound rules make a call, I see "Unknow or Blocked" on the called phone.
I try to go to SIP menu or Inbound rules but I don't see where I can enter the default number for phones that haven't inbound rules.
And for complicate all of that, in SIP menu the Account ID are not a phone line but a login with characters (like: company1-ox453)
I say that because in "inbound rules" menu I see at the top of page : "default CLID company1-ox453".
It's possible to override the CLID in specific menu or in next firmware ?
Thank you for your help.

Hi Stephane,
The account ID will be used as the default caller ID for the internal phones which don't have inbound rule. In your case, the acount ID is with characters, so the external phones can't display it as caller ID. UC320W can't override the CLID.
Best regards,
Wendy Yang

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Cisco 2911 Voice Gateway SIP PSTN Calls Fail

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From: <sip:[email protected]>:tag=6166CDC4-882
To: <sip:[email protected]>
Shawn C. Smith

i have same problem my cucm ip is 192.168.200.53
my Voice Gateway is SIP by ip 192.168.200.86 for internal
and 172.29.7.94
and my SIP Server is 10.208.9.69
if its oky can yuo take a look at my problem please
this is the syslog from debug
May 30 20:19:34.284: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:192.168.200.53:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
Session-Expires: 1800
P-Asserted-Identity: "Aysar Mohamed" <sip:[email protected]>
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
May 30 20:19:34.284: //-1/B5A494800000/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=2217156
   ----- ccCallInfo IE subfields -----
   cisco-ani=2217156
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=90555769123
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
May 30 20:19:34.288: //-1/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x30CF41D4, Call Info(
   Calling Number=2217156,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=90555769123(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=465
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 1
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832953048,feature_id:85
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=90555769123(TON=Unknown, NPI=Unknown))
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
   Event=0x2B82D890
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 90555769123
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetContext:
   Context=0x2ABC2E44
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 465 with tag 0 to app "_ManagedAppProcess_Default"
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=802, Params=0x2ABC19D4, Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
   In: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
   Out: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Destination Pattern=9T, Called Number=0555769123, Digit Strip=FALSE
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0555769123(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=Aysar Mohamed
   Account Number=2217156, Final Destination Flag=TRUE,
   Guid=B5A49480-0001-0000-0000-11C535C8A8C0, Outgoing Dial-peer=802
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=2217156
   ----- ccCallInfo IE subfields -----
   cisco-ani=2217156
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=0555769123
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x30CF41D4, Interface Type=3, Destination=, Mode=0x0,
   Call Params(Calling Number=2217156,(Calling Name=Aysar Mohamed)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0555769123(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=802, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 2
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832952824,feature_id:86
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccCallSetContext:
   Context=0x2ABC1984
May 30 20:19:34.292: //465/B5A494800000/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=802
May 30 20:19:34.292: //466/B5A494800000/CCAPI/cc_api_call_proceeding:
   Interface=0x30CF41D4, Progress Indication=NULL(0)
May 30 20:19:34.292: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
May 30 20:19:34.292: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1401481174
Contact: <sip:[email protected]:5060>
Call-Info: <sip:172.29.7.94:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
May 30 20:19:34.300: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
CSeq: 101 INVITE
Content-Length: 0
May 30 20:19:34.612: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Contact: <sip:[email protected]:5060;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Content-Length: 328
Content-Type: application/sdp
v=0
o=- 17192647 17192647 IN IP4 10.208.9.69
s=SBC call
c=IN IP4 10.208.9.69
t=0 0
m=audio 39910 RTP/AVP 8 0 102 102 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 AMR/8000
a=rtpmap:102 AMR/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5
a=fmtp:116 0-15
a=fmtp:18 annexb=yes
May 30 20:19:34.612: %SIP-3-UNSUPPORTED: Unsupported ptime value
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=466,
   Caps(Codec=0x2, Fax Rate=0x2, Vad=0x1,
   Modem=0x0, Codec Bytes=160, Signal Type=2)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
   Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
   Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
   Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
   Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event=170, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event=98, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
   Interface=0x30CF41D4, Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1),
   Cause Value=0
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
   Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
   Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1), Cause Value=0
   Voice Call Send Alert=FALSE, Call Entry(Alert Sent=FALSE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
   Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccConferenceCreate:
   (confID=0x30C11410, callID1=0x1D1, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
   (confID=0x30C11410, callID2=0x1D2, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   Conference Id=0x30C11410, Call Id1=465, Call Id2=466, Tag=0x0
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
May 30 20:19:34.616: ccConferenceCreate: ret1=0, codecMask1=2, bytes1=160, negot1=0, dtmf1=0
                    ret2=0, codecMask2=2, bytes2=160, negot2=1, dtmf2=6,
                    tx_dynamic_pt1=0, rx_dynamic_pt1=0, codec_mode1=0, params_bitmap1 =0
                    tx_dynamic_pt2=8, rx_dynamic_pt2=8, codec_mode2=0, params_bitmap2 =0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   delay media to slow start case, codec negotation is not done
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_api_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
   Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/B5A494800000/CCAPI/cc_api_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x16, Destination Call Id=466)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x16, Destination Call Id=465)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_process_notify_bridge_done:
   Conference Id=0x16, Call Id1=465, Call Id2=466
May 30 20:19:34.616: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 233
v=0
o=CiscoSystemsSIP-GW-UserAgent 2639 5276 IN IP4 192.168.200.86
s=SIP Call
c=IN IP4 192.168.200.86
t=0 0
m=audio 18288 RTP/AVP 8 0 19
c=IN IP4 192.168.200.86
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Reason: Q.850;cause=127;text="interworking unspecified"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
   Cause Value=41, Interface=0x30CF41D4, Call Id=466
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=41, Retry Count=0)
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccCallReleaseResources:
   release reserved xcoding resource.
May 30 20:19:34.680: //466/B5A494800000/CCAPI/ccCallSetAAA_Accounting:
   Accounting=0, Call Id=466
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccConferenceDestroy:
   Conference Id=0x16, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
   Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=466,
   Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: :cc_free_feature_vsa freeing 31A5D9F0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: vsacount in free is 1
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=41
Content-Length: 0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=465,
   Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: :cc_free_feature_vsa freeing 31A5DAD0
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: vsacount in free is 0
May 30 20:19:36.044: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.7.94:5060 SIP/2.0
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0
May 30 20:19:36.048: //467/8DAABF6C8278/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>;tag=739BBC-1CE2
Date: Fri, 30 May 2014 20:19:36 GMT
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 446
v=0
o=CiscoSystemsSIP-GW-UserAgent 3496 1601 IN IP4 172.29.7.94
s=SIP Call
c=IN IP4 172.29.7.94
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.7.94
m=image 0 udptl t38
c=IN IP4 172.29.7.94
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
My SIP GW internal ip address is 192.168.200.86
and the Public IP is : 172.29.7.94
My CUCM is 192.168.200.53
my GW Config is :
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 3
rule 1 /^9/ /\1/
voice translation-rule 4
rule 4 /^22217/ /7/
rule 5 /^2217/ /7/
rule 6 /^022217/ /7/
rule 7 /^0122217/ /7/
voice translation-rule 5
rule 1 /^5/ /905/
rule 2 /^1/ /901/
rule 3 /^2/ /902/
rule 4 /^3/ /903/
rule 5 /^4/ /904/
rule 6 /^6/ /906/
rule 7 /^7/ /907/
rule 8 /^8/ /908/
rule 10 /^00/ /900/
rule 11 /'+'/ /900/
voice translation-profile OUT
translate called 3
voice translation-profile REDIAL
translate calling 5
voice translation-profile SIP-NEW
translate called 4
application
service mva http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
service ccm http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
license udi pid CISCO2921/K9 sn FCZ164960G0
hw-module pvdm 0/0
hw-module pvdm 0/1
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.200.86 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/1
ip address 172.29.7.94 255.255.255.252
duplex auto
speed auto
ip http server
ip http access-class 23
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip route 0.0.0.0 0.0.0.0 192.168.200.1
ip route 10.208.9.0 255.255.255.0 172.29.7.93
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 192.168.200.53 identifier 1 priority 1 version 7.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register NAGHI-MTP
dspfarm profile 2 mtp
codec g711alaw
maximum sessions hardware 25
associate application SCCP
dial-peer voice 802 voip
description ** SIP TO STC **
translation-profile outgoing OUT
destination-pattern 9T
session protocol sipv2
session target ipv4:10.208.9.69:5060
session transport udp
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay sip-notify rtp-nte sip-kpml
no vad
dial-peer voice 811 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 812 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 813 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 814 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 815 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 816 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 817 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 818 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
Please i need ur help ASAP

Restoring Default Call Type on N85

I have an issue with regard to the default call type on my Nokia N85. When I have a VOIP service configured, the call type defaults to "Internet Call", and this is particularly annoying since I can no longer use my dialing shortcuts. The "Default Call Type" setting does not exist on my phone; I was hoping that an upgrade to the latest firmware would correct this, but it has not. I am using the latest software release of Nokia's VOIP implementation, as well. Does anyone know of a backend way in which I can restore the default call type back to GSM? I would greatly appreciate this. I've had to remove the VOIP service from my phone and resort to using my VOIP service's access number in order to make VOIP calls.
Thank you.
Daryl

Maybe it has something to do with the VoIP service you have installed. I have used the SIP VoIP Settings application and there the default call type is GSM voice call after changing the setting in the VoIP Service menu. When none of the VoIP services is set as default and dialling e.g. 123456 on the keypad, the call goes out as a normal voice call.
It might be worth checking that there's no other VoIP service set as default.

CUCM: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls

Hi Team,
we are running CUCM 9.1(2a),
we have integrated Third Party SIP Phone(Avaya 1230 SIP Phone) with CUCM,
Issue: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls, we are able to see only the dailed Number,
When "A" calls to "B", "A" can see only the dailed number of "B" but not the "Caller ID"
Regards
Ananthakumar

Are A and B both Avaya phones?
So it looks like you're not seeing the alerting name/connected name getting updated then? Do you have alerting names configured on the directory numbers? Might need to take a look at the SIP messaging to see if the alerting name/connected name is being sent to the Avaya phones and maybe they just aren't displaying it. Might just be something that needs to be tweaked in the 46xxsettings.txt file.

Reject sip/h323 calls by IP?

i have a few sip/h323 providers. I have also enabled sip/h323 on my as5400xm(this is for my asterisk server). Since i'm using these providers, i have to put their IP in my access-list. my concern is, since my gateway is accepting sip/h323 calls. what if these provider send the calls to my gateway? so i was thinking of a way to restrict this. It could be as simple as tweaking the access-list. but I don't know. Please help.
here's how i have my access-list setup:
access-list 101 permit tcp host 10.10.10.10 any
access-list 101 permit udp host 10.10.10.10 any
access-list 101 permit udp any any range 16384 32767
access-list 101 deny tcp any any
access-list 101 deny udp any any
Thanks in advance

Ah, so you just want to restrict VoIP calls from L3 addresses other than your provider?
That's just a simple ACL to open up traffic to your SIP ITSP's IP external addresses, and block anything else.
You can get what IPs and ports are used by your provider, but here is what you need open on the Cisco side inbound for an inbound ACL on a WAN interface:
UDP - ITSP address:ITSP SIP Port to External interface:5060 - For SIP signaling
ITSP address:ITSP RTP Port Range - External interface:16384-32767 - RTP traffic
ITSP's port range could be anything between 1024-65535. SIP usually comes from UDP/5060 from the ITSP, but doesn't have to. Verify with them, or look at a SIP debug or packet capture to verify.
The implicit deny will take care of everything else.

Any SIP/ VoIP Expert? Echo of own voice in SIP conference call?

Hi,
I am developing an VoIP application using Jain-SIP library. My problem is when my application makes a SIP conference call, I can hear my own voice in the phone, however when making a SIP normal call, it won't happen. How do you disable the echo of own voice in a conference call? Thanks for your reading and appreciate your help!

Hello giliadw,
Don’t wait for a built in function because that’s no where soon to be seen in near future. You should certainly contact a hosted PBX provider they definitely provide VoIP solutions for mobiles. You can simply function all the SIP tools through their application. I can recommend you my PBX VoIP provider The RealPBX, they have a decent phone application and services

Default Call Completion Date

When we TECO a PM work order created from a maintenance plan we specify the TECO date for the order and the call completion date for the plan. We've configured the default order TECO date to be the basic finish date on the order. The call completion date is defaulted to the plan date for that call. We'd like to specify a default call completion date other than the plan date but we're not sure the default call completion date is configurable and if it is, where to change it?
Thanks for the help!
David Macindoe

David,
No its not configurable.
If you are using ECC6, then you may be able to use the Enhancement Framework options to develop a solution - ask your ABAP team.
PeteA

BlackBerry Z10 - Skype as unwanted default Calling App

Hi all,
I would like to kindly ask you for help with Skype for Z10.
Current issue is that Skype sets itself as default calling application which I don’t want.
Example: when someone sends me an SMS with phone number and I will click on it, Z10 will automatically open Skype, instead of standard phone application. How can I disable it or repair it, please?
Note: Uninstalling Skype is not a solution what I'm looking for.
Thanks
Quido
8310 @ 5.0 -> 9700 @ 6.0 -> 9900 @ 7.1 -> Z10 @ 10.3.1
T-Mobile CZ, BES

This is a problem with the current version if Skype. Signing out if Skype is the only option I'm aware of, aside from units talking it. You can also press and hold a number, then select "call".

What is default Call barring code,pin code1,pin co...

I have newly bought new e51.But user guide mention only default device code,not other codes. Anyone please tell me what is default call barring code,pin code 1&2.I have also got advanced device lock already in memory card.But its lock code also not mentioned in user guide.

In case of advanced device when i installed it,it asked me to enter old code if I want to change password.when I selected some folders for lock it ask me the password now to unlock it.But yet I have not set any new password as I do not know old default password. Hey friend,also please tell me default call barring password.

SIP incoming call, won't work (CME)

Hi all,
I'm facing a weird problem and the sip-provider can't help. I suppose there is a problem with the dial-peer/translation-rule but I can't figure it out...
There is a CME (c2800nm-ipvoice-mz.124-11.XW10.bin, CME Version 4.2(0)) with a
SIP trunk.
Outgoing calls are working (DID).
Incoming calls (all DID) are ringing on the same internal number.
The situation:
- external call on 0815440097 is ringing on the internal nr. 296 (should be 297)
- external call on 0815440096 is ringing on the internal nr. 296
Here the config:
================================
voice service voip
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
no update-callerid
voice translation-rule 40
rule 2 /$.*$/ /9\1/
voice translation-rule 190
rule 1 /^0$.*$/ /\1/
rule 2 /^9$.*$/ /\1/
voice translation-rule 191
rule 2 /296/ /0815440096/
rule 3 /297/ /0815440097/
voice translation-rule 192
rule 2 /^0815440097/ /297/
rule 3 /^0815440096/ /296/
voice translation-profile TP_IN_SIP
translate calling 40
translate called 192
voice translation-profile TP_OUT_SIP
translate calling 191
translate called 190
dial-peer voice 2000 voip
description *** SIP-TRUNK (IN/OUT) ***
translation-profile incoming TP_IN_SIP
translation-profile outgoing TP_OUT_SIP
destination-pattern 0.T
b2bua
session protocol sipv2
session target dns:sip12.e-fon.ch
session transport udp
incoming called-number 0815440096
dtmf-relay rtp-nte
codec g711alaw
no vad
sip-ua
credentials username 0815440096 password 7 xxxx realm sip12.e-fon.ch
keepalive target dns:sip12.e-fon.ch
authentication username 0815440096 password 7 xxxx
calling-info pstn-to-sip from number set 0815440096
no remote-party-id
retry invite 2
retry response 2
retry bye 2
retry register 2
retry options 1
registrar dns:sip12.e-fon.ch expires 69
sip-server dns:sip12.e-fon.ch
reason-header override
connection-reuse
host-registrar
sh sip-ua register status
Line                              peer        expires(sec) registered
================================ ========== ============ ==========
0815440096                        20005       18            yes
Here the CCSIP MESSAGE debug (looks ok):
(call from 0000000000 to 0815440097)
===============================
Mar 8 21:55:10.469 METD: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:212.55.198.132;lr=on;ftag=as00cd0e7f>
Via: SIP/2.0/UDP 212.55.198.132;branch=z9hG4bK49ff.2d35e30a71291ffe3895b39164900f36.0
Via: SIP/2.0/UDP 212.55.198.134:5061;branch=z9hG4bK1cb84749;rport=5061
Max-Forwards: 69
From: "0000000000" <sip:[email protected]:5061>;tag=as00cd0e7f
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: e-fon
Date: Thu, 08 Mar 2012 20:55:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-IPCONNECT: 0815440097
X-Number: 0815440097
Content-Type: application/sdp
Content-Length: 415
v=0
o=root 770254981 770254981 IN IP4 212.55.198.134
s=Asterisk PBX 1.6.1.20
c=IN IP4 212.55.198.134
t=0 0
m=audio 11886 RTP/AVP 8 9 111 3 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Mar 8 21:55:10.481 METD: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.55.198.132;branch=z9hG4bK49ff.2d35e30a71291ffe3895b39164900f36.0,
Via:SIP/2.0/UDP 212.55.198.134:5061;branch=z9hG4bK1cb84749;rport=5061
From: "0000000000" <sip:[email protected]:5061>;tag=as00cd0e7f
To: <sip:[email protected]:5060>
Date: Thu, 08 Mar 2012 20:55:10 GMT
Call-ID: [email protected]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
Here is the VOICE DIAL-PEER debug (call from 0000000000 to 0815440097):
=============================================
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=0815440096, Called Number=0815440096, Peer Info
Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=0815440096
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20005
2: Dial-peer Tag=2000
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=0000000000, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2000
Mar 8 22:00:09.502 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=0000000000, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.502 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2000
Mar 8 22:00:09.502 METD: //-1/8647979A82E1/DPM/dpAssociateIncomingPeerCore:
Calling Number=0000000000, Called Number=0815440096, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.502 METD: //-1/8647979A82E1/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2000
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=0815440096, Called Number=0815440096, Peer Info
Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=0815440096
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20006
2: Dial-peer Tag=2000
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=0815440096, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ANSWER; Incoming Dial-peer=2000
Mar 8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersCore:
Calling Number=, Called Number=0815440096, Peer Info
Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=0815440096
Mar 8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Mar 8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20006
2: Dial-peer Tag=2000
show dial-peer voice summary:
dial-peer hunt 0
AD                                    PRE PASS                OUT
TAG    TYPE MIN OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT
PORT
555    voip up   up             555                0 syst loopback:rtp
20001 pots up   up             296$               0                          50/0/1
20002 pots up   up             297$               0                          50/0/2
2000   voip up   up             0.T                0 syst dns:sip12.e-fon.ch
20005 pots up   up             0815440096$        0                     50/0/150
20006 pots up   up             0815440097$        9                     50/0/2
voip translation debugging (call from 0794142975 to 0815440097):
=========================================
Mar 8 22:35:26.145 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFCA0; count=1
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFCA0; count=0
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: number=0794142975 type=unknown plan=unknown numbertype=calling
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/sed_subst: Successful substitution; pattern=0794142975 matchPattern=(.*) replacePattern=9\1 replaced pattern=90794142975
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: xlt_number=90794142975 xlt_type=unknown xlt_plan=unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-called
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-called number not found
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: number=0815440096 type=unknown plan=unknown numbertype=called
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=3
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: No match found
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: No match: number=0815440096 type=unknown plan=unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFCA0; count=1
Mar 8 22:35:26.153 METD: //-1/73E51DB2834F/RXRULE/regxrule_dp_translate: No profile found in peer 20005 for outgoing direction
Mar 8 22:35:26.153 METD: //-1/73E51DB2834F/RXRULE/regxrule_dp_translate: calling_number=90794142975 calling_octet=0x0
        called_number=0815440096 called_octet=0x0
        redirect_number= redirect_type=0 redirect_plan=0        redirect_PI=-1 redirect_SI=-1
Mar 8 22:35:26.181 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFCA0; count=2
Thanks,
Norbert

Hi Alex,
Thank you for the reply.
After changing the "incoming called-number" I got the same output.
The weird think is, why the dial-peer debug shows the 0815440096 number, despite the right "to: number" in the SIP-Message.
Is there a problem with the "voice service voip" or "sip-ua"?
on the voice translation debug I see:
Match Rule=DP_MATCH_TO_URI; URI=sip:0815440097
Match Rule=DP_MATCH_FROM_URI; URI=sip:0819262424
But I guess the translation rule is maching this one:
Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=0815440096
So how can the voice translation rule be set to map the entry DP_MATCH_TO_URI; URI=sip:0815440097
Thanks for the help.
Regards,
Norbert
voip translation debugging (call from 0819262424 to 0815440097):
===================================================
Mar 9 07:45:16.371 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFAFC; count=1
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFAFC; count=0
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: number=0819262424 type=unknown plan=unknown numbertype=calling
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/sed_subst: Successful substitution; pattern=0819262424 matchPattern=(.*) replacePattern=9\1 replaced pattern=90819262424
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: xlt_number=90819262424 xlt_type=unknown xlt_plan=unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-called
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-called number not found
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: number=0815440096 type=unknown plan=unknown numbertype=called
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 3 in ruleset 192
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 3 in ruleset 192
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/sed_subst: Successful substitution; pattern=0815440096 matchPattern=^0815440096 replacePattern=296 replaced pattern=296
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: xlt_number=296 xlt_type=unknown xlt_plan=unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFAFC; count=1
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_dp_translate: No profile found in peer 20001 for outgoing direction
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_dp_translate: calling_number=90819262424 calling_octet=0x0
        called_number=296 called_octet=0x0
        redirect_number= redirect_type=0 redirect_plan=0        redirect_PI=-1 redirect_SI=-1
Mar 9 07:45:16.379 METD: //-1/439ABF97847F/RXRULE/regxrule_vp_translate: No profile found in voice port or trunk group for outgoing direction
Mar 9 07:45:16.379 METD: //-1/439ABF97847F/RXRULE/regxrule_vp_translate: calling_number=90819262424 calling_octet=0x0
        called_number=296 called_octet=0x0
        redirect_number= redirect_type=0 redirect_plan=0
Mar 9 07:45:18.195 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFAFC; count=2
debug voice dialpeer detail
=====================
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=0815440096, Expanded String=0815440096, Calling Number=0815440096T
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=2000 Is Matched
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=20005 Is Matched
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ANSWER; Calling Number=0819262424
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=0819262424T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Result=-1
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ORIGINATE; Calling Number=0819262424
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=0819262424T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
   Result=Success(0); Incoming Dial-peer=2000 Is Matched
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ANSWER; Calling Number=0819262424
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=0819262424T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Result=-1
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ORIGINATE; Calling Number=0819262424
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=0819262424T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
   Result=Success(0); Incoming Dial-peer=2000 Is Matched
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_REQUEST_URI; URI=sip:[email protected]:5060
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
   Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_TO_URI; URI=sip:[email protected]:5060
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
   Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_FROM_URI; URI=sip:[email protected]:5061
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
   Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=0815440096
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
   Dial String=0815440096, Expanded String=0815440096, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
   Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ANSWER; Calling Number=0819262424
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=0819262424T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
   Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ORIGINATE; Calling Number=0819262424
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=0819262424T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/MatchNextPeer:
   Result=Success(0); Incoming Dial-peer=2000 Is Matched
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=296, Expanded String=296, Calling Number=296T
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=20001 Is Matched
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ANSWER; Calling Number=296
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=296T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Result=-1
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ORIGINATE; Calling Number=296
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=296T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
   Result=Success(0); Incoming Dial-peer=20001 Is Matched
Mar 9 07:49:25.584 METD: //-1/D8245087848D/DPM/dpMatchCore:
   Dial String=296, Expanded String=296, Calling Number=
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.584 METD: //-1/D8245087848D/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=20001 Is Matched
Mar 9 07:49:25.588 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=90819262424, Expanded String=90819262424, Calling Number=
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.588 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Result=-1
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=0819262424, Expanded String=0819262424, Calling Number=
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=2000 Is Matched
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ANSWER; Calling Number=296
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=296T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Result=-1
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ORIGINATE; Calling Number=296
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=296T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
   Result=Success(0); Incoming Dial-peer=20001 Is Matched
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=296, Expanded String=296, Calling Number=
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=20001 Is Matched
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=90819262424, Expanded String=90819262424, Calling Number=
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Result=-1
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=90819262424, Expanded String=90819262424, Calling Number=
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Result=-1
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=296, Expanded String=296, Calling Number=
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=20001 Is Matched

CME SIP Phone Calls in one-way (inside local network)

Hello everyone, first time here, need a little help.
I'm having some trouble to find a solution to the following problem.
Recently I've installed CME 9.1 using the router 2921. Most of the phones are SIPs, model 3905 (around 20 of them), with the last firmware updated.
Some users are complaining one way audio issue in internal calls, from a extension to another (only in sip phones)
With Wireshark capture I could see that RTP packets are being sent and receive by the router and not directly trough the phones. Is this normal in CME? When a call with problems occours (one way audio) there is no audio in one way, but router still sends confort noise packets.
Here is my config.
Thanks for any help.
Martin
##################################################################################33
System returned to ROM by power-on
System restarted at 11:29:23 BR Tue Jan 29 2013
System image file is "flash0:c2900-universalk9-mz.SPA.152-4.M2.bin"
Last reload type: Normal Reload
Last reload reason: power-on
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 10.3.245.1 port 5060
max-dn 60
max-pool 70
load ATA-187 ATA187.9-2-3-1
load 3905 CP3905.9-2-1-0
authenticate realm all
timezone 17
time-format 24
date-format D/M/Y
tftp-path flash:
file text
create profile sync 0094230880392697
network-locale U1
user-locale U1 load /CME-locale-pt_BR-Portuguese-8.8.2.5.tar
ntp-server 10.3.244.7 mode directedbroadcast
voice register dn 1
number 9006
name Sala_Reuniao_02
label Sala de Reuniao 2
voice register dn 2
number 9007
name Sala_Reuniao_03
voice register dn 3
number 9008
name Sala Reuniao 04
voice register pool 1
id mac 8478.ACE6.09A2
type 3905
number 1 dn 1
template 1
codec g711ulaw
voice register pool 2
id mac 8478.ACE6.0573
type 3905
number 1 dn 2
codec g711ulaw
voice register pool 3
id mac 5897.1ECD.8F8D
type 3905
number 1 dn 3
codec g711ulaw
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.220
encapsulation dot1Q 220
ip address 10.3.245.1 255.255.255.0
ip helper-address 10.3.244.71
h323-gateway voip bind srcaddr 10.3.245.1
telephony-service
max-ephones 5
max-dn 5 no-reg both
ip source-address 10.3.245.1 port 2000
timeouts interdigit 5
timeouts busy 12
system message XXXXXXXX
cnf-file location flash:
cnf-file perphone
user-locale U2 load CME-locale-pt_BR-Portuguese-8.8.2.5.tar
user-locale 2 PT
network-locale U2
load 7925 CP7925G-1.4.1SR1.LOADS
load 6941 SCCP69xx.9-2-1-0.loads
time-zone 17
time-format 24
date-format dd-mm-yy
max-conferences 8 gain -6
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 9001
ephone 1
mac-address D867.D9E6.F57F
ephone-template 1
type 6941
button 1:1

Hi ,
We have upgarded the the firmware to the 3905.9-2-2ES2 , but show voice register pool phone-load still shows the old firmware, but the phoen itself is showing the new upgraded version on the dsiplay ...any advice is highly appricated,
ADM-CME9#show voice register pool phone-load
Pool Device Name     Current-Version             Previous-Version
==== =============== =========================== ===========================
1    SEP7081053DE72F Cisco/SPA502G-7.4.8a
3    SEP34BDC8C6C412 Cisco-CP3905/9.2.1
4    SEP34BDC8C64561 Cisco-CP3905/9.2.1
5    SEP54781AE1F531 Cisco-CP3905/9.2.1
6    SEP54781AE171D2 Cisco-CP3905/9.2.1
10   SEP54781AE1F544 Cisco-CP3905/9.2.1
15   SEP1CE6C77323CD Cisco-CP3905/9.2.1
16   SEP58971E282A23 Cisco-CP3905/9.2.1
17   SEP58971E2822A8 Cisco-CP3905/9.2.1
19   SEP1CE6C77321F3 Cisco-CP3905/9.2.1
30   SEP54781AE171E2 Cisco-CP3905/9.2.1
31   SEP54781AE16FD4 Cisco-CP3905/9.2.1
32   SEP54781AE16F2F Cisco-CP3905/9.2.1
33   SEP54781A1C77FD Cisco-CP3905/9.2.1
34   SEP54781A1C77DC Cisco-CP3905/9.2.1
35   SEP54781AE17527 Cisco-CP3905/9.2.1
36   SEP54781AE17766 Cisco-CP3905/9.2.1
37   SEP54781AE1731A Cisco-CP3905/9.2.1
38   SEP54781AE08B8D Cisco-CP3905/9.2.1
39   SEP54781AE123B1 Cisco-CP3905/9.2.1

Redirect SIP Trunk calls to FXO port

Hi,
This is the scenario. There are 3 branches, two of them are Cisco Call Manager Express and one of them is Elastix-based.
So, as the image explains, the three branches have SIP trunks fully operational. The branches are in different cities, so the numbers structure changes. In city A it begins with 2, in B begins with 3 and in C begins with 4. Every POTS number is a 7 digit number (2XXXXXX, 3XXXXXX, 4XXXXXX). And every user, in every branch, have a 4 digit number beginning with the city code (2XXX, 3XXX, 4XXX).
But, every time city A wants to make a call to a POTS number in city B, it goes across the A´s FXO line. So it charges a inter-city cost to the call.
The client wants that every time a city A user wants to call a POTS number in city B, goes over the SIP trunk to city B and use the FXO on the city B call manager.
I have made a pattern for city A. So, everytime the user dials 3XXXXXX, it does not use the city A´s FXO, but it goes to the branch in city B.
What do I have to do now in branch B´s Call Manager Express to redirect that call to a local FXO?
Thanks in advanced!
Regards
PS. There is a diagram of the topology. Want to do what the red line is doing.

In this situation I would do an answer-address based on ANI so you are specifically identifying your site A and then just piggy back off the local FXO out.
So assuming you are sending just 4 digits over the SIP for each site:
Dial-peer voice X voip
answer-address "blah"
protocol sipv2
...(whatever else you need to configure in these dots)
At this point your CME at site B will take the call see that it is destined for a POTs line and it should send it out whatever local dial-peer you have setup for that site when they dial out to the PSTN locally.
EDIT:
Then again, you probably already have a general incoming dial-peer, the above design would just be specific for your site A and isn't really needed.

DX70/80 and external SIP video calls

We have a BE6K running CUCM 10 and Collab Edge. Our main conf room has an MX300. We have a couple rich media licenses.
We are looking at the new DX70/80. From what I understand they will work fine for video calling within our cluster. But will they work with Collab Edge and external SIP video conferencing to a third party? Do I just need to make sure I have enough rich media licenses? Anything else?
Thanks!

Any Telepresence Endpoint/Codec running TC7.0.1 or later firmware is enabled for MRA with Expressway.
Additionally, yes, you need the RMS licenses for your Expressway-Core and Edge to traverse the call for B2B calling as well as a SIP Dial-Plan in CUCM to send the calls to Expressway.
Source: http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-1/Mobile-Remote-Access-via-Expressway-Deployment-Guide-X8-1-1.pdf

Default caller name for a given phone number

If you have multiple contacts that have a phone number in common, is it possible to decide which of those contacts is the default display name for that number?
For example, I have an entry for my wife. I also have an entry for "ICE" (In Case of Emergency). They're essentially the same. When my wife calls or texts me, the display says "ICE." I want it to say "Wife."
Is this possible?

Nope.
BTW, I am curious, do you have a passcode lock on your phone? If so, doesn't that pretty much make any ICE contact pointless since they cannot access the phone?
Understand your desire to have it there though. Also, there are apps out there (think called ICE) that can let you put a icon on your screen that says ICE and then link that to a number.
Message was edited by: DaVBMan

OLC Reject for audio codec in a H.323-SIP B2B call involving CUCM & Expressways

In the setup, CUCM is the centralized call controller and expressways are used for firewall traversal. CUCM & expressways are correctly as per the Cisco document. A SIP call from a public internet endpoint to CUCM registered MX300 G2 is successfully connected with video & audio. But a H.323 call from public internet endpoint to the MX300 G2 in CUCM is failing. It is seen in the logs that there is a OpenLogicalChannel reject for the audio codec. I tried with both G711 & G729 from the public endpoint, but the result is the same. Has anyone here faced similar problems. Is there anything that I have look at. Thanks in advance.

Hi Martin,
Thanks for the response. I can look at the firewall. But I would like to understand how does firewall can be a factor here when there is a clear OLC Reject. Is it not confirms that the packets are indeed received and there are some issues in opening the logical channel.
As firewall is not in my scope, I need to make clear reason for the other team to look at.
Thanks.

UC320 SIP Default Caller ID

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