Outbound caller-id name
I have a UC540W with a Time Warner PRI. When I call someone outside the office the caller ID shows up as the first number in my DID block. Is it possible to be changed so it display my company name?
Darren,
Thanks for the reply, I tired renaming the description of the phones but it still appears to use the name fields when sending the caller ID. For example even when I changed the description to "Company Name" it still appeared as Board Room to external users.
I have not tried the strip name as of yet, as I am trying to avoid doing CLI config if possible (to keep the CCA supported), but am curious if I will need to apply this to all of my dial-peers related to my PRI. I am assuming so but would be great if you could confirm or if you had any other thoughts on how to block this.
Thanks again for the quick response
Similar Messages
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485 Ambiguous - Outbound Calls Only
I'm having some issues with the 485 Ambigous Error, but on Outbound calls only. I've read several blog posts and was able to solve this issue for incoming calls, but have yet to find a solution for outbound calls.
I have two phone numbers: XXX-XXX-5232 and XXX-XXX-3081. All of my users are configured in Lync with +1XXXXXX5232;ext=XXXXX
XXX-XXX-5232 is set to normalize to +1XXXXXX5232;ext=53999 which is a Lync User that is setup with Team Call to ring multiple other extensions. I'm not aware of any AutoAttendant or Response group that is configured with just +1XXXXXX5232.
Again, inbound calling is working just fine and outbound calls work as expected, they just report the 485 Ambiguous for each call.
Thanks in advance for any help you can provide.Yes, the call is routed to the Gateway. In fact, the call completes successfully. Here is the trace. Hopefully it will be readable:
12:17:03.513 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288602 ) ---- Incoming SIP Message from 10.188.0.18:61275 to SIPInterface #0 ---- [Time: 11:17:03]
12:17:03.543 : 10.188.0.19 : NOTICE : INVITE sip:[email protected];user=phone SIP/2.0
FROM: "VRRL - Kitchen"<sip:+1XXXXXX5232;[email protected];user=phone>;epid=621B6C2CF5;tag=aeb33cd2cc
TO: <sip:[email protected];user=phone>
CSEQ: 32959 INVITE
CALL-ID: 14fe8790-50ef-476a-80ac-e57061c0a2af
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 10.188.0.18:61275;branch=z9hG4bK7f7f1664
CONTACT: <sip:VRRL-SBA.cdol.int:5067;transport=Tls;ms-opaque=86db4f0fd1133b15>
CONTENT-LENGTH: 552
SUPPORTED: 100rel
USER-AGENT: RTCC/4.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
ALLOW: ACK
Allow: CANCEL,BYE,INVITE,PRACK,UPDATE
v=0
o=- 885 1 IN IP4 10.188.0.18
s=session
c=IN IP4 10.188.0.18
b=CT:1000
t=0 0
m=audio 53854 RTP/AVP 97 101 13 0 8
c=IN IP4 10.188.0.18
a=tcap:1 RTP/SAVP
a=pcfg:1 t=1
a=rtcp:53855
a=label:Audio
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ssoH69QQto9/wyQyDEbtGezAe4zuH4ulyHNtUfRT|2^31|1:1
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:Eas2Y5diRZ5HKgxFHpLLTdr8EWMmERj6ZGLjf8LO|2^31
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
[Time: 11:17:03]
12:17:03.573 : 10.188.0.19 : NOTICE : ( sip_stack)(2288604 ) new AcSIPCallAPI created - #276 [Time: 11:17:03]
12:17:03.593 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288605 ) | | new GetNewSIPCall created - #517 [Time: 11:17:03]
12:17:03.603 : 10.188.0.19 : NOTICE : ( lgr_stk_mngr)(2288606 ) Resource StackSession <#276> Allocated [Time: 11:17:03]
12:17:03.613 : 10.188.0.19 : NOTICE : ( sip_stack)(2288607 ) TlsTransportObject#57::CheckForConnectionPersistent - Opening persistent connection with proxy: 10.188.0.18:61275 [Time: 11:17:03]
12:17:03.613 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288608 ) | |(SIPTU#517)INVITE State:Idle() [Time: 11:17:03]
12:17:03.623 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288609 ) DNSResolver::HandleARecordQuery - Host:VRRL-SBA.cdol.int resolved in external table [Time: 11:17:03]
12:17:03.633 : 10.188.0.19 : NOTICE : ( sip_stack)(2288610 ) (SIPTU#517) HandleResolutionSuccessEV: Domain name VRRL-SBA.cdol.int was successfully resolved to IP: 10.188.0.18 [Time: 11:17:03]
12:17:03.643 : 10.188.0.19 : NOTICE : ( sip_stack)(2288611 ) SIPCall(#517) changes state from Idle to Invited [Time: 11:17:03]
12:17:03.653 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288612 ) | | | #276:SIP_DNS_RESOLVED_EV(14fe8790-50ef-476a-80ac-e57061c0a2af)
[Time: 11:17:03]
12:17:03.663 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288613 ) | | | #276:SIP_SETUP_EV(14fe8790-50ef-476a-80ac-e57061c0a2af)
[Time: 11:17:03]
12:17:03.673 : 10.188.0.19 : NOTICE : ( lgr_call)(2288614 ) (#276) Call Allocated. [Time: 11:17:03]
12:17:03.673 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288615 ) SIPStackSession::HandleStackSetupEV - NEWCALL: SrcPN=0 [Time: 11:17:03]
12:17:03.683 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288616 ) <SESSION #276> SendToCall - event: NEW_CALL_EV m_Call#276 [Time: 11:17:03]
12:17:03.693 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288617 ) | | #276:NEW_CALL_EV:(14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.703 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288618 ) | | #276:Call changing states from:IdleState to:NewCallState_IP2Tel [Time:
11:17:03]
12:17:03.713 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288619 ) ServicesMngr::GetEndPoint PhoneNum = 402XXX0899
[Time: 11:17:03]
12:17:03.713 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288620 ) GetTrunkGroupId- TrunkGroup:1 found DstNum:402XXX0899 DstPfx:* SrcNum:+1XXXXXX5232 SrcPfx:* SrcIp:abc0012 SrcIpPfx:10.188.0.18 [Time: 11:17:03]
12:17:03.723 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288621 ) QueryOnHookPortStatus (ChannelNum=0), status = 1 Polarity = 0 [Time: 11:17:03]
12:17:03.733 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288622 ) Current trunks status: [Time: 11:17:03]
12:17:03.743 : 10.188.0.19 : NOTICE : ( lgr_num)(2288623 ) PhoneNumber::RemovePrefix - Number change from +1XXXXXX5232 to 1XXXXXX5232 [Time: 11:17:03]
12:17:03.753 : 10.188.0.19 : NOTICE : ( lgr_call)(2288624 ) Call::SetCoderListForCall #276 Found 2 Common Coders For Call [Time: 11:17:03]
12:17:03.763 : 10.188.0.19 : NOTICE : ( lgr_call)(2288625 ) <Call #276> Coder g711Ulaw64k20 : 20 [Time: 11:17:03]
12:17:03.763 : 10.188.0.19 : NOTICE : ( lgr_call)(2288626 ) <Call #276> Coder g711Alaw64k20 : 20 [Time: 11:17:03]
12:17:03.773 : 10.188.0.19 : NOTICE : ( lgr_profiling)(2288627 ) <Call 276> Profiled<Tel=0,Ip=0>: JBMinDel=10 JBOptF=10 EEarlyM=1 FaxTM=1 IPDS=46 IsFaxU=2 PI2IP=-1 SigIPDF=40 CNGMode=0 DTMFUsed=0 NSEMode=0 PlayRBTone2IP=1
RBUdpPort=0 RTPRD=0 SCE=0 VxxTT=2 Dst2Rdrt=0 DTMFVol=20 ECE=1 ECurDis=0 EDigDel=0 ERevP=0 FHPer=700 InG=32 MWIA=0 MWID=0 VVol=32 ReorderTime=255 DIDWink=0 2StageDial=0 DiscOnBusyT=1 DiscOnBrok=1 DPInd=255 AGC=0 NLP=0 [Time: 11:17:03]
12:17:03.783 : 10.188.0.19 : NOTICE : ( lgr_call)(2288628 ) | | #276GetNextUI:GlobalUI=442334516, mACAddrLsb=3257879 [Time: 11:17:03]
12:17:03.793 : 10.188.0.19 : NOTICE : ( lgr_call)(2288629 ) | | #276GetNextUI:GlobalUI=442334517 [Time: 11:17:03]
12:17:03.803 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288630 ) | #0:NEW_CALL_EV : (14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.813 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288631 ) EndPoint::MediaResourceList::AllocateMediaIpPortsByMediaRealmID Perform NEW allocation of Media ports for RealmIndex(0) port(6220) current allocations
are:(1) [Time: 11:17:03]
12:17:03.813 : 10.188.0.19 : NOTICE : ( sip_stack)(2288632 ) SIPSDPSession#276 - Changing state from SIP_MEDIA_IDLE to SIP_MEDIA_OFFERED [Time: 11:17:03]
12:17:03.823 : 10.188.0.19 : NOTICE : ( sip_stack)(2288633 ) <BaseSIP SDPSESSION #276> UpdateChosenMediaByCN - CN as Remote 1 [Time: 11:17:03]
12:17:03.833 : 10.188.0.19 : NOTICE : ( sip_stack)(2288634 ) <BaseSIP SDPSESSION #276> UpdateChosenMediaByCN - Force silence suppression on chosen coder, because remote & local support CN [Time: 11:17:03]
12:17:03.843 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288635 ) | |(SIPTU#517)TRYING_REQ State:Invited(14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.853 : 10.188.0.19 : NOTICE : ( sip_stack)(2288636 ) New SIPMessage created - #58 [Time: 11:17:03]
12:17:03.863 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288637 ) ---- Outgoing SIP Message to 10.188.0.18:61275 from SIPInterface #0 ---- [Time: 11:17:03]
12:17:03.873 : 10.188.0.19 : NOTICE : SIP/2.0 100 Trying
Via: SIP/2.0/TLS 10.188.0.18:61275;branch=z9hG4bK7f7f1664
From: "VRRL - Kitchen"<sip:+1XXXXXX5232;[email protected];user=phone>;epid=621B6C2CF5;tag=aeb33cd2cc
To: <sip:[email protected];user=phone>;tag=1c274616087
Call-ID: 14fe8790-50ef-476a-80ac-e57061c0a2af
CSeq: 32959 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.20A.045.006
Content-Length: 0
[Time: 11:17:03]
12:17:03.883 : 10.188.0.19 : NOTICE : ( sip_stack)(2288639 ) Resource SIPMessage deleted - #58 [Time: 11:17:03]
12:17:03.883 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288640 ) SIPStackSession::HandleStackSetupEV - SETUP: SrcPN=0 [Time: 11:17:03]
12:17:03.893 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288641 ) <SESSION #276> SendToCall - event: SETUP_EV m_Call#276 [Time: 11:17:03]
12:17:03.903 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288642 ) | | #276:SETUP (TO:402XXX0899, FROM:+1XXXXXX5232):(14fe8790-50ef-476a-80ac-e57061c0a2af)
[Time: 11:17:03]
12:17:03.913 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288643 ) | | #276:Call changing states from:NewCallState_IP2Tel to:InitiatedState_IP2Tel
[Time: 11:17:03]
12:17:03.923 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288644 ) | #0:SETUP_EV : (14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.933 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288645 ) UpdateChannelParams, Channel 0
[Time: 11:17:03]
12:17:03.943 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288646 ) #0:PSOSBoardInterface::ConfigureFaxModemChannelParams FAXTransportType=3 Modem configuration VxxTransportType=2 not allowed, forced to 3
[Time: 11:17:03]
12:17:03.953 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288647 ) #0:ConfigFaxModemChannelParams NSEMode=0, CNGDetMode=0, FAXTranType=3, VxxTranType=3, VoiceVol= 0, DTMFVol=-11, InGain=0, RTPRedDepth=0, ECE=1, SCE=3, ECNlpMode=0,
DJBufMinDelay=10, DJBufOptFac=10, Result=1) [Time: 11:17:03]
12:17:03.963 : 10.188.0.19 : NOTICE : ( lgr_psbrdif)(2288648 ) Turn ringer ON for channel 0 [Time: 11:17:03]
12:17:03.973 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288649 ) | #0:FXO Seize Line [Time: 11:17:03]
12:17:03.973 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288650 ) | #0:ALERT_EV (send) : (14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.983 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288651 ) | | #276:ALERT_EV:(14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.993 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288652 ) | | #276:Call changing states from:InitiatedState_IP2Tel to:AlertingState_IP2Tel
[Time: 11:17:03]
12:17:04.003 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288653 ) | | | #276:ALERT_EV(14fe8790-50ef-476a-80ac-e57061c0a2af)
[Time: 11:17:03]
12:17:04.013 : 10.188.0.19 : NOTICE : ( sip_stack)(2288654 ) New SIPMessage created - #93 [Time: 11:17:03]
12:17:04.013 : 10.188.0.19 : NOTICE : ( sip_stack)(2288655 ) SIPSDPSession#276 - Changing state from SIP_MEDIA_OFFERED to SIP_MEDIA_COMPLETED [Time: 11:17:03]
12:17:04.023 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288656 ) DtmfCapNegotiationAlgorithm :: TxDtmfMethod = DTMF_RFC2833_SUPPORTED [Time: 11:17:03]
12:17:04.033 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288657 ) DtmfCapNegotiationAlgorithm :: TxRtpRfc2833Payload = 101 [Time: 11:17:03]
12:17:04.043 : 10.188.0.19 : NOTICE : ( lgr_stk_ses)(2288658 ) <SESSION #276> SendToCall - event: DTMF_CONTROL_EV m_Call#276 [Time: 11:17:03]
12:17:04.053 : 10.188.0.19 : NOTICE : ( lgr_flow)(2288659 ) | | #276:DTMF_CONTROL_EV:(14fe8790-50ef-476a-80ac-e57061c0a2af) [Time:
11:17:03]
12:17:04.063 : 10.188.0.19 : NOTICE : SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 10.188.0.18:61275;branch=z9hG4bK7f7f1664
From: "VRRL - Kitchen"<sip:+1XXXXXX5232;[email protected];user=phone>;epid=621B6C2CF5;tag=aeb33cd2cc
To: <sip:[email protected];user=phone>;tag=1c274616087
Call-ID: 14fe8790-50ef-476a-80ac-e57061c0a2af
CSeq: 32959 INVITE
Contact: <sip:[email protected]:5067;transport=tls>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Require: 100rel
RSeq: 1
Server: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.20A.045.006
Content-Type: application/sdp
Content-Length: 254
v=0 -
Outbound call failing with cause code 57
Hi,
our outbound calls to some numbers getting failed with cause code 57 and in ccsip messages i am getting 403 forbidden.
i tried to change the payload type to 97 which was 98 but no success.
the called number is 9-8955900
the calling number is 8062300
can any one help me..
the ccsip messages and ccapi inout debug is..
509022: *Jan 8 14:23:20.513: :FEATURE_VSA attributes are: feature_name:0,featur e_time:1255632752,feature_id:53127
509023: *Jan 8 14:23:20.513: //678454/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPr ivate:
SPI Call Setup Request Is Success; Interface Type=9, FlowMode=1
509024: *Jan 8 14:23:20.513: //678454/xxxxxxxxxxxx/CCAPI/ccCallSetContext:
Context=0x4AC5D304
509025: *Jan 8 14:23:20.517: //678454/xxxxxxxxxxxx/CCAPI/cc_api_call_connected:
Interface=0x48D4E620, Data Bitmask=0x0, Progress Indication=NULL(0),
Connection Handle=0
509026: *Jan 8 14:23:20.517: //678454/xxxxxxxxxxxx/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
509027: *Jan 8 14:23:20.537: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.12.190:5060;branch=z9hG4bK6aded64034252
From: "Asif CIPC" <sip:[email protected]>;tag=2524413~70e9433b-1d79-44ae-9a16- 09a52be377c5-22878662
To: <sip:[email protected]>
Date: Wed, 08 Jan 2014 14:02:15 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
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 2031538944-0000065536-0000027607-3188500672
Session-Expires: 1800
P-Asserted-Identity: "Asif CIPC" <sip:[email protected]>
Remote-Party-ID: "Asif CIPC" <sip:[email protected]>;party=calling;screen=yes; privacy=off
Contact: <sip:[email protected]:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 240
v=0
o=CiscoSystemsCCM-SIP 2524413 1 IN IP4 192.168.12.190
s=SIP Call
c=IN IP4 192.168.33.5
t=0 0
m=audio 17706 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
509028: *Jan 8 14:23:20.553: //-1/7916D3000000/CCAPI/cc_api_display_ie_subfield s:
cc_api_call_setup_ind_common:
cisco-username=3064
----- ccCallInfo IE subfields -----
cisco-ani=3064
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=8955900
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
509029: *Jan 8 14:23:20.553: /
ASICO-DAM#/-1/7916D3000000/CCAPI/cc_api_call_setup_ind_common:
Interface=0x48667600, Call Info(
Calling Number=3064,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=T RUE,
Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALS E), Call Id=678455
509030: *Jan 8 14:23:20.553: //-1/7916D3000000/CCAPI/ccCheckClipClir:
In: Calling Number=3064(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pre sentation=Allowed)
509031: *Jan 8 14:23:20.553: //-1/7916D3000000/CCAPI/ccCheckClipClir:
Out: Calling Number=3064(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pr esentation=Allowed)
509032: *Jan 8 14:23:20.553: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509033: *Jan 8 14:23:20.553: :cc_get_feature_vsa malloc success
509034: *Jan 8 14:23:20.553: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509035: *Jan 8 14:23:20.553: cc_get_feature_vsa count is 13
509036: *Jan 8 14:23:20.553: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509037: *Jan 8 14:23:20.553: :FEATURE_VSA attributes are: feature_name:0,featur e_time:1255634768,feature_id:53128
509038: *Jan 8 14:23:20.553: //678455/7916D3000000/CCAPI/cc_api_call_setup_ind_ common:
Set Up Event Sent;
Call Info(Calling Number=3064(TON=Unknown, NPI=Unknown, Screening=User, Passe d, Presentation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown))
509039: *Jan 8 14:23:20.557: //678455/7916D3000000/CCAPI/cc_process_call_setup_ ind:
Event=0x48EE6200
509040: *Jan 8 14:23:20.557: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 8955900
509041: *Jan 8 14:23:20.561: //678455/7916D3000000/CCAPI/ccCallSetContext:
Context=0x476E35D0
509042: *Jan 8 14:23:20.561: //678455/7916D3000000/CCAPI/cc_process_call_setup_ ind:
>>>>CCAPI handed cid 678455 with tag 1 to app "_ManagedAppProcess_Default"
509043: *Jan 8 14:23:20.561: //678455/7916D3000000/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
509044: *Jan 8 14:23:20.565: //678455/7916D3000000/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=20, Params=0x476E4AE0, Progress Indication=NULL(0)
509045: *Jan 8 14:23:20.565: //678455/7916D3000000/CCAPI/ccCheckClipClir:
In: Calling Number=8062301(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
509046: *Jan 8 14:23:20.565: //678455/7916D3000000/CCAPI/ccCheckClipClir:
Out: Calling Number=8062301(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
509047: *Jan 8 14:23:20.565: //678455/7916D3000000/CCAPI/ccCallSetupRequest:
Destination Pattern=.T, Called Number=8955900, Digit Strip=FALSE
509048: *Jan 8 14:23:20.565: //678455/7916D3000000/CCAPI/ccCallSetupRequest:
Calling Number=8062301(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pres entation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Asif CIPC
Account Number=3064, Final Destination Flag=TRUE,
Guid=7916D300-0001-0000-0000-6BD7BE0CA8C0, Outgoing Dial-peer=20
509049: *Jan 8 14:23:20.565: //678455/7916D3000000/CCAPI/cc_api_display_ie_subf ields:
ccCallSetupRequest:
cisco-username=3064
----- ccCallInfo IE subfields -----
cisco-ani=8062301
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=8955900
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
509050: *Jan 8 14:23:20.569: //678455/7916D3000000/CCAPI/ccIFCallSetupRequestPr ivate:
Interface=0x48667600, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=8062301,(Calling Name=Asif CIPC)(TON=Unknown, NPI= Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=20 , Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Appl ication Call Id=)
509051: *Jan 8 14:23:20.569: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509052: *Jan 8 14:23:20.569: :cc_get_feature_vsa malloc success
509053: *Jan 8 14:23:20.569: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509054: *Jan 8 14:23:20.569: cc_get_feature_vsa count is 14
509055: *Jan 8 14:23:20.569: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509056: *Jan 8 14:23:20.569: :FEATURE_VSA attributes are: feature_name:0,featur e_time:1255629840,feature_id:53129
509057: *Jan 8 14:23:20.569: //678456/7916D3000000/CCAPI/ccIFCallSetupRequestPr ivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
509058: *Jan 8 14:23:20.573: //678456/7916D3000000/CCAPI/ccCallSetContext:
Context=0x476E4A90
509059: *Jan 8 14:23:20.573: //678455/7916D3000000/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=20
509060: *Jan 8 14:23:20.577: //678456/7916D3000000/CCAPI/cc_api_call_proceeding :
Interface=0x48667600, Progress Indication=NULL(0)
509061: *Jan 8 14:23:20.585: //678456/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172..XX.XX.XX:5060;branch=z9hG4bK67CD11AE
Remote-Party-ID: "Asif CIPC" <sip:[email protected]>;party=calling;screen=ye s;privacy=off
From: "Asif CIPC" <sip:[email protected]>;tag=EA475228-24AE
To: <sip:[email protected]>
Date: Wed, 08 Jan 2014 14:23:20 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2031538944-0000065536-0000027607-3188500672
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF Y, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1389191000
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 274
v=0
o=CiscoSystemsSIP-GW-UserAgent 5380 1731 IN IP4 172..XX.XX.XX
s=SIP Call
c=IN IP4 172..XX.XX.XX
t=0 0
m=audio 19502 RTP/AVP 18 101
c=IN IP4 172..XX.XX.XX
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
509062: *Jan 8 14:23:20.589: //678455/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.12.190:5060;branch=z9hG4bK6aded64034252
From: "Asif CIPC" <sip:[email protected]>;tag=2524413~70e9433b-1d79-44ae-9a16- 09a52be377c5-22878662
To: <sip:[email protected]>
Date: Wed, 08 Jan 2014 14:23:20 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
509063: *Jan 8 14:23:20.605: //678456/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172..XX.XX.XX:5060;branch=z9hG4bK67CD11AE
Call-ID: [email protected]
From: "Asif CIPC"<sip:[email protected]>;tag=EA475228-24AE
To: <sip:[email protected]>
CSeq: 101 INVITE
Content-Length: 0
509064: *Jan 8 14:23:20.677: //678456/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172..XX.XX.XX:5060;branch=z9hG4bK67CD11AE
Record-Route: <sip:10.205.20.50:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Asif CIPC"<sip:[email protected]>;tag=EA475228-24AE
To: <sip:[email protected]>;tag=sbc0804k7h28358
CSeq: 101 INVITE
Reason: Q.850;cause=57;text="bearer capability not authorized"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0
509065: *Jan 8 14:23:20.677: //678456/7916D3000000/CCAPI/cc_api_call_disconnect ed:
Cause Value=57, Interface=0x48667600, Call Id=678456
509066: *Jan 8 14:23:20.677: //678456/7916D3000000/CCAPI/cc_api_call_disconnect ed:
Call Entry(Responsed=TRUE, Cause Value=57, Retry Count=0)
509067: *Jan 8 14:23:20.681: //678455/7916D3000000/CCAPI/ccCallReleaseResources :
release reserved xcoding resource.
509068: *Jan 8 14:23:20.681: //678456/7916D3000000/CCAPI/ccCallSetAAA_Accountin g:
Accounting=0, Call Id=678456
509069: *Jan 8 14:23:20.681: //678456/7916D3000000/CCAPI/ccCallDisconnect:
Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect C ause=57)
509070: *Jan 8 14:23:20.681: //678456/7916D3000000/CCAPI/ccCallDisconnect:
Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
509071: *Jan 8 14:23:20.681: //678456/7916D3000000/CCAPI/cc_api_call_disconnect _done:
Disposition=0, Interface=0x48667600, Tag=0x0, Call Id=678456,
Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
509072: *Jan 8 14:23:20.685: //678456/7916D3000000/CCAPI/cc_api_call_disconnect _done:
Call Disconnect Event Sent
509073: *Jan 8 14:23:20.685: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509074: *Jan 8 14:23:20.685: :cc_free_feature_vsa freeing 4AD76408
509075: *Jan 8 14:23:20.685: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509076: *Jan 8 14:23:20.685: vsacount in free is 13
509077: *Jan 8 14:23:20.685: //678455/7916D3000000/CCAPI/ccCallDisconnect:
Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect C ause=0)
509078: *Jan 8 14:23:20.689: //678455/7916D3000000/CCAPI/ccCallDisconnect:
Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
509079: *Jan 8 14:23:20.693: //678455/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/TCP 192.168.12.190:5060;branch=z9hG4bK6aded64034252
From: "Asif CIPC" <sip:[email protected]>;tag=2524413~70e9433b-1d79-44ae-9a16- 09a52be377c5-22878662
To: <sip:[email protected]>;tag=EA475298-106E
Date: Wed, 08 Jan 2014 14:23:20 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=57
Content-Length: 0
509080: *Jan 8 14:23:20.693: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172..XX.XX.XX:5060;branch=z9hG4bK67CD11AE
From: "Asif CIPC" <sip:[email protected]>;tag=EA475228-24AE
To: <sip:[email protected]>;tag=sbc0804k7h28358
Date: Wed, 08 Jan 2014 14:23:20 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
509081: *Jan 8 14:23:20.709: //678454/xxxxxxxxxxxx/CCAPI/ccCallDisconnect:
Cause Value=0, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Ca use=0)
509082: *Jan 8 14:23:20.709: //678454/xxxxxxxxxxxx/CCAPI/ccCallDisconnect:
Cause Value=0, Call Entry(Responsed=TRUE, Cause Value=0)
509083: *Jan 8 14:23:20.709: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.12.190:5060;branch=z9hG4bK6aded64034252
From: "Asif CIPC" <sip:[email protected]>;tag=2524413~70e9433b-1d79-44ae-9a16- 09a52be377c5-22878662
To: <sip:[email protected]>;tag=EA475298-106E
Date: Wed, 08 Jan 2014 14:02:15 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0
509084: *Jan 8 14:23:20.713: //678455/7916D3000000/CCAPI/cc_api_call_disconnect _done:
Disposition=0, Interface=0x48667600, Tag=0x0, Call Id=678455,
Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
509085: *Jan 8 14:23:20.717: //678455/7916D3000000/CCAPI/cc_api_call_disconnect _done:
Call Disconnect Event Sent
509086: *Jan 8 14:23:20.717: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509087: *Jan 8 14:23:20.717: :cc_free_feature_vsa freeing 4AD77748
509088: *Jan 8 14:23:20.717: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509089: *Jan 8 14:23:20.717: vsacount in free is 12
509090: *Jan 8 14:23:20.721: //678454/xxxxxxxxxxxx/CCAPI/cc_api_call_disconnect _done:
Disposition=0, Interface=0x48D4E620, Tag=0x0, Call Id=678454,
Call Entry(Disconnect Cause=0, Voice Class Cause Code=0, Retry Count=0)
509091: *Jan 8 14:23:20.721: //678454/xxxxxxxxxxxx/CCAPI/cc_api_call_disconnect _done:
Call Disconnect Event Sent
509092: *Jan 8 14:23:20.721: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509093: *Jan 8 14:23:20.721: :cc_free_feature_vsa freeing 4AD76F68
509094: *Jan 8 14:23:20.721: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509095: *Jan 8 14:23:20.721: vsacount in free is 11
509096: *Jan 8 14:23:20.725: //-1/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivat e:
Interface=0x48D4E620, Interface Type=9, Destination=0.0.0.0, Mode=0x0,
Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screeni ng=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=, FinalDestinationFlag=FALSE, Outgoing Dial-peer=0, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Appl ication Call Id=)
509097: *Jan 8 14:23:20.725: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509098: *Jan 8 14:23:20.725: :cc_get_feature_vsa malloc success
509099: *Jan 8 14:23:20.725: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509100: *Jan 8 14:23:20.729: cc_get_feature_vsa count is 12
509101: *Jan 8 14:23:20.729: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509102: *Jan 8 14:23:20.729: :FEATURE_VSA attributes are: feature_name:0,featur e_time:1255632752,feature_id:53130
509103: *Jan 8 14:23:20.729: //678457/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPr ivate:
SPI Call Setup Request Is Success; Interface Type=9, FlowMode=1
509104: *Jan 8 14:23:20.729: //678457/xxxxxxxxxxxx/CCAPI/ccCallSetContext:
Context=0x4AC5D1C4
509105: *Jan 8 14:23:20.729: //678457/xxxxxxxxxxxx/CCAPI/cc_api_call_connected:
Interface=0x48D4E620, Data Bitmask=0x0, Progress Indication=NULL(0),
Connection Handle=0
509106: *Jan 8 14:23:20.729: //678457/xxxxxxxxxxxx/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
509107: *Jan 8 14:23:20.753: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.12.190:5060;branch=z9hG4bK6adee3e42a498
From: "Asif CIPC" <sip:[email protected]>;tag=2524415~70e9433b-1d79-44ae-9a16- 09a52be377c5-22878662
To: <sip:[email protected]>
Date: Wed, 08 Jan 2014 14:02:15 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
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 2031538944-0000065536-0000027608-3188500672
Session-Expires: 1800
P-Asserted-Identity: "Asif CIPC" <sip:[email protected]>
Remote-Party-ID: "Asif CIPC" <sip:[email protected]>;party=calling;screen=yes; privacy=off
Contact: <sip:[email protected]:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 240
v=0
o=CiscoSystemsCCM-SIP 2524415 1 IN IP4 192.168.12.190
s=SIP Call
c=IN IP4 192.168.33.5
t=0 0
m=audio 17932 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
509108: *Jan 8 14:23:20.769: //-1/7916D3000000/CCAPI/cc_api_display_ie_subfield s:
cc_api_call_setup_ind_common:
cisco-username=3064
----- ccCallInfo IE subfields -----
cisco-ani=3064
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=8955900
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
509109: *Jan 8 14:23:20.769: //-1/7916D3000000/CCAPI/cc_api_call_setup_ind_comm on:
Interface=0x48667600, Call Info(
Calling Number=3064,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=T RUE,
Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALS E), Call Id=678458
509110: *Jan 8 14:23:20.769: //-1/7916D3000000/CCAPI/ccCheckClipClir:
In: Calling Number=3064(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pre sentation=Allowed)
509111: *Jan 8 14:23:20.773: //-1/7916D3000000/CCAPI/ccCheckClipClir:
Out: Calling Number=3064(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pr esentation=Allowed)
509112: *Jan 8 14:23:20.773: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509113: *Jan 8 14:23:20.773: :cc_get_feature_vsa malloc success
509114: *Jan 8 14:23:20.773: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509115: *Jan 8 14:23:20.773: cc_get_feature_vsa count is 13
509116: *Jan 8 14:23:20.773: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509117: *Jan 8 14:23:20.773: :FEATURE_VSA attributes are: feature_name:0,featur e_time:1255634768,feature_id:53131
509118: *Jan 8 14:23:20.773: //678458/7916D3000000/CCAPI/cc_api_call_setup_ind_ common:
Set Up Event Sent;
Call Info(Calling Number=3064(TON=Unknown, NPI=Unknown, Screening=User, Passe d, Presentation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown))
509119: *Jan 8 14:23:20.773: //678458/7916D3000000/CCAPI/cc_process_call_setup_ ind:
Event=0x48EE6200
509120: *Jan 8 14:23:20.777: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 8955900
509121: *Jan 8 14:23:20.777: //678458/7916D3000000/CCAPI/ccCallSetContext:
Context=0x476D5190
509122: *Jan 8 14:23:20.777: //678458/7916D3000000/CCAPI/cc_process_call_setup_ ind:
>>>>CCAPI handed cid 678458 with tag 1 to app "_ManagedAppProcess_Default"
509123: *Jan 8 14:23:20.777: //678458/7916D3000000/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
509124: *Jan 8 14:23:20.781: //678458/7916D3000000/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=20, Params=0x476DCE60, Progress Indication=NULL(0)
509125: *Jan 8 14:23:20.781: //678458/7916D3000000/CCAPI/ccCheckClipClir:
In: Calling Number=8062301(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
509126: *Jan 8 14:23:20.781: //678458/7916D3000000/CCAPI/ccCheckClipClir:
Out: Calling Number=8062301(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
509127: *Jan 8 14:23:20.785: //678458/7916D3000000/CCAPI/ccCallSetupRequest:
Destination Pattern=.T, Called Number=8955900, Digit Strip=FALSE
509128: *Jan 8 14:23:20.785: //678458/7916D3000000/CCAPI/ccCallSetupRequest:
Calling Number=8062301(TON=Unknown, NPI=Unknown, Screening=User, Passed, Pres entation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Asif CIPC
Account Number=3064, Final Destination Flag=TRUE,
Guid=7916D300-0001-0000-0000-6BD8BE0CA8C0, Outgoing Dial-peer=20
509129: *Jan 8 14:23:20.785: //678458/7916D3000000/CCAPI/cc_api_display_ie_subf ields:
ccCallSetupRequest:
cisco-username=3064
----- ccCallInfo IE subfields -----
cisco-ani=8062301
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=8955900
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
509130: *Jan 8 14:23:20.785: //678458/7916D3000000/CCAPI/ccIFCallSetupRequestPr ivate:
Interface=0x48667600, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=8062301,(Calling Name=Asif CIPC)(TON=Unknown, NPI= Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=8955900(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=20 , Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Appl ication Call Id=)
509131: *Jan 8 14:23:20.785: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509132: *Jan 8 14:23:20.785: :cc_get_feature_vsa malloc success
509133: *Jan 8 14:23:20.785: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509134: *Jan 8 14:23:20.785: cc_get_feature_vsa count is 14
509135: *Jan 8 14:23:20.785: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
509136: *Jan 8 14:23:20.785: :FEATURE_VSA attributes are: feature_name:0,featur e_time:1255629840,feature_id:53132
509137: *Jan 8 14:23:20.789: //678459/7916D3000000/CCAPI/ccIFCallSetupRequestPr ivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
509138: *Jan 8 14:23:20.789: //678459/7916D3000000/CCAPI/ccCallSetContext:
Context=0x476DCE10
509139: *Jan 8 14:23:20.789: //678458/7916D3000000/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=20
509140: *Jan 8 14:23:20.793: //678459/7916D3000000/CCAPI/cc_api_call_proceeding :
Interface=0x48667600, Progress Indication=NULL(0)
509141: *Jan 8 14:23:20.801: //678459/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172..XX.XX.XX:5060;branch=z9hG4bK67CE197B
Remote-Party-ID: "Asif CIPC" <sip:[email protected]>;party=calling;screen=ye s;privacy=off
From: "Asif CIPC" <sip:[email protected]>;tag=EA475304-26C5
To: <sip:[email protected]>
Date: Wed, 08 Jan 2014 14:23:20 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2031538944-0000065536-0000027608-3188500672
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF Y, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1389191000
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 274
v=0
o=CiscoSystemsSIP-GW-UserAgent 9218 9584 IN IP4 172..XX.XX.XX
s=SIP Call
c=IN IP4 172..XX.XX.XX
t=0 0
m=audio 16868 RTP/AVP 18 101
c=IN IP4 172..XX.XX.XX
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
509142: *Jan 8 14:23:20.805: //678458/7916D3000000/SIP/Msg/ccsipDisplayMsg:
Sent:
509163: *Jan 8 14:23:20.945: //678458/7916D3000000/CCAPI/cc_api_call_disconnect _done:
Call Disconnect Event Sent
509164: *Jan 8 14:23:20.945: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509165: *Jan 8 14:23:20.945: :cc_free_feature_vsa freeing 4AD77748
509166: *Jan 8 14:23:20.945: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
509167: *Jan 8 14:23:20.945: vsacount in free is 12
509168: *Jan 8 14:23:32.517: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172..XX.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKhs8ak5845ch42p7cff35k1ap3T02677
Call-ID: isbchh12748fcsk155w58p151kks36fww24s@SoftX3000
From: <sip:172..XX.XX.XX:5060>;tag=sbc0806pa8fp7w7
To: <sip:172..XX.XX.XX>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0
509169: *Jan 8 14:23:32.525: //678460/495AB05FB187/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKhs8ak5845ch42p7cff35k1ap3T02677
From: <sip:172..XX.XX.XX:5060>;tag=sbc0806pa8fp7w7
To: <sip:172..XX.XX.XX>;tag=EA4780CC-582
Date: Wed, 08 Jan 2014 14:23:32 GMT
Call-ID: isbchh12748fcsk155w58p151kks36fww24s@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF Y, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 170
v=0
o=CiscoSystemsSIP-GW-UserAgent 8937 2437 IN IP4 172..XX.XX.XX
s=SIP Call
c=IN IP4 192.168.33.5
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.33.5
u all
and the config is
voice service voip
ip address trusted list
ipv4 172.XX.XX.XX 255.255.255.255
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
no supplementary-service h225-notify cid-update
redirect ip2ip
h323
session transport udp
h245 tunnel disable
sip
session transport tcp
rel1xx disable
registrar server expires max 3600 min 3500
transport switch udp tcp
redirect contact order best-match
asymmetric payload full
g729 annexb-all
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
and the dial-peer through the calls will go out is
dial-peer voice 20 voip
description ***TO-OUT***
translation-profile outgoing OUT-SIP
destination-pattern .T
progress_ind progress enable 8
rtp payload-type cisco-codec-fax-ack 112
rtp payload-type nte 97
session protocol sipv2
session target ipv4:10.205.20.50:5060
session transport udp
voice-class codec 1
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vadHi Nadeem,
our setup is like
vpn link
call manager ------head office(router)----------------------------------brach router(gateway)------------------ip phone(branch)
The call flow is
sip
IP Phone---->brach GW-------------->call manager------------------>branch GW------------------->ITSP
there is no subscriber at branch side, so the outbound call should travel to call manager at head office and then get exit from branch gateway to ITSP through sip line. -
I know that the default outbound caller id is always wireless caller but I had a question. If I set it to my name would that cost me additional money a month?
You can try
Phone Number
800-922-0204
Talk to a Human
Press #, then 6, then 0 when it asks for your number. EXISTING CUSTOMERS press 4, then say "Customer Service" then wait and enter last four of social security number. -
SPA112 FW bug confirmed: registration broken during and after outbound call
After receiving and configuring my new SPA112, everything seemed to work out just fine: registration to the SIP proxy, inbound calls, etc. However, at some point I discovered that outbound calls would typically only work for a certain amount of time, after which the connection to the callee would be lost, just like the my registration to the SIP proxy. In the SPA112's Web interface, the device reports "Registration State: Failed" in such cases. Only a reboot of the device will result in a successful registration again. Firmware version: 1.3.5(004).
To diagnose the problem, I enabled both syslog and debug logging and started capturing and analyzing what is going wrong:
[Before call]
SIP.REGISTER
SIP.TRYING
SIP.UNAUTHORIZED (with nonce)
SIP.REGISTER
SIP.TRYING
SIP.OK
Also the debug logging reports "LOCAL3.DEBUG: [0]RegOK. NextReg in X".
So far, so good.
[During/after call]
We see two (periodic) registration sequences:
SIP.REGISTER
SIP.TRYING
SIP.UNAUTHORIZED (with nonce)
LOCAL3.DEBUG: SIP_tsClientEventProc(event: 28)
LOCAL3.DEBUG: SIP_regTsEventProc(event: 28)
LOCAL3.DEBUG: SIP_tsClientEventProc(event: 28)
LOCAL3.DEBUG: SIP_regTsEventProc(event: 28)
LOCAL3.DEBUG: SIP_regTsEventProc(event: 28)
LOCAL3.DEBUG: SIP_tsCreateClient()
SIP.REGISTER
SIP.TRYING
LOCAL3.DEBUG: SIP_tsClientEventProc(event: 28)
LOCAL3.DEBUG: SIP_regTsEventProc(event: 28)
SIP.OK
LOCAL3.DEBUG: SIP_tsClientEventProc(event: 28)
LOCAL3.DEBUG: SIP_regTsEventProc(event: 28)
LOCAL3.DEBUG: [0]RegOK. NextReg in X
Which is OK and the same as before the call. This one is however shortly followed by the following (second registration sequence of) debug messages:
LOCAL3.DEBUG: SIP_tsClientEventProc(event: 4)
LOCAL3.DEBUG: tryAltIp_delay()
LOCAL3.DEBUG: RSE_DEBUG: getting alternate from domain:<proxy>
LOCAL3.DEBUG: SIP_regTsEventProc(event: 4)
LOCAL3.DEBUG: SIP_regTsEventProc(event: 32)
LOCAL3.DEBUG: 2. RSE:GetServerAddrErr(<proxy>,0)=-101
LOCAL3.DEBUG: SIP_tsCreateClient(), 1781, uiTmrF=1600, SIP_TMR_F_INIT=1600
LOCAL3.DEBUG: [0]Reg Addr Change(0) 29dfc552:5060->0:5060
LOCAL3.DEBUG: tpError at [SIP_tsCreateClient:1817]
These two registration sequences (regular SIP messages at the one hand and the debug messages at the other hand) are alternating, and the succesfull registration is followed by the failing registration in a short time, so most typical state is "Failed".
Any suggestion on what I could try to solve the issue is welcome!I'm able to reproduce the problem in my test lab.
It seems that device can't fetch IP address of the SIP proxy from it's internal cache during active call (GetServerAddrErr error -101). And the issue doesn't solve in full even after the calls end.
It is not networking issue between device and DNS server as no DNS query has been issued (the IP address is cached from previous query that has been responded successfully).
As a workaround I configured proxy IP instead of name.
It work.
It seems we hit (another) firmware bug. -
Cisco Jabber for Windows in Extend and Connect mode and making outbound calls
Hi guys,
I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up.
However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs.
After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number.
Has anyone got this working or can provide some guidance?
Thanks.Hi guys,
I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up.
However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs.
After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number.
Has anyone got this working or can provide some guidance?
Thanks. -
July 3, 2013 :
(1) Spoke to Toyota Technical Support today about 2013 Toyota entune, had them gooogle "wiki BlackBerry Z10" had them, then, to click 3rd one down on google listing : " BlackBerry - Wikipedia, the free encyclopedia " had them, then, see picture of my BlackBerry Z10 and under it on the right side 5th item down see and read to me : " Units shipped 200 million "
Toyota states they are working on their 2013 Toyota entune included for 36 months which integrates iPhone V and Samsung Galaxy Android but no BlackBerry to Navigation, SmartPhone usage, e-mail to Z10, read e-mail while driving Z10, and Contacts with address to extend 100 maximum limitation into 2013 Toyota Navigation on-board silly memory, USB support to Z10 with the USB plug provided, POI points of interest on Navigation where you click it and it dials the phone number on record of every business selected by 200 categories gas dentist doctor pet doctor groceries bakery restaurant shopping mall motel camping grounds RV hook-up etc.
(2) T-Mobile currently supports 10.1.0.2006 but I have loaded on previous BlackBerry products bootleg updated s/w releases which worked perfectly well and solved many of my previous issues on the 10 or so previous BlackBerry products I’ve owned since BlackBerry became the choice for users where security is the main purpose for business users, not mp3 listening to music for Apple which is just an over-priced under-technology product with lots of apps all of which I am complaining about in this post.
I’d like to do that again now, and see if any of these issues we all are having are being worked on in the least by BlackBerry. Where can I get instructions of where to download whatever is the bootleg people know does fix at least 1 of these issues, please ?
(3) In-coming calls show only phone number. Never had a BlackBerry like that, ever. Understand this is a known bug, while others waste your time with swipe down from Phone App and Click Settings and Click one of the 13 settings in there such as SMART DIALING and do something with the 3 area codes local calls where I live and something with National Number Length also not explained where it can be corrected doing that. I need to see the caller-id name and number, some folks refer to as Deluxe Caller-ID. Number and name, not just the number. No idea who is calling me, ever.
(4) Take in-coming call and press the number in the PHONE LOG on Phone App, and it calls the customer back. I want to save the number to CONTACTS and either choose a contact to add it to, or have multiple listings for the same contact where I see one has a number and one just the e-mail I added. I am still not satisfied with the add new contact every time I add via e-mail address when I already have the e-mail address and want to see it. But, at least I can add contacts to the several thousand I deal with and at least have them in there twice. That’s a start.
(5) Visual Voice Mail, I am now paying for at T-Mobile. And, I have programmed it on on the Z10. What is Visual vm ? If I have it, where do I see it ? What is it the Z10 provides on visual vm ?
(6) There must be an updated manual from the lackluster manual I downloaded when I upgraded from the BlackBerry Bold 9780 to Z10 which goes into a lot more detail than the one I downloaded, printed and went through to get to where I’m at, which is
Still after many months now, having a Z10 phone which :
(1) does nothing with 2013 Toyota entune ?
(2) must have a bootleg solid s/w release I can load with instructions ?
(3) in-coming caller phone number and Name, not just # ?
(4) Phone Log App click last in-coming call and Add to Contacts ?
(5) Visual vm turned on both carrier and settings and not seeing it ?
(6) latest manual on Z10 which someone surely has updated to address these issues
Anyone else having these issues items (1) through (6) ?
Give Thumbs-Up if helps you, or say Solution Accepted, or say Like.
Solved!
Go to Solution.My pleasure. And, I'm a guy. I'll look through my settings when I have a chance later today. Have a good one.
-
Automatic navigation after "END" button depending on in- or outbound call
Hi all.
the requirement is:
In the case of inbound calls - the agent starts working on an item of the inbox and clicks on "END" when finished - the navigation should automatically bring him back to the inbox.
But if he starts with the account identification (outbound calls) then he should automatically jump back to account identification screen when clicking on "END".
I already got the Intent driven interaction working according to these steps. But now the navigation brings me ALWAYS back to the inbox.
1) Define a rule policy in the Rule Profile for context "Intent driven interaction":
If
Current Event Equals InteractionEnded-Interaction ended
Then
Navigate To ( Navigation Object type:Action = IC WebClient - Inbox:Display )
2) Release the rule using the "more" button and clicking on "relase draft rule" (the icon changes from yellow to green)
3) On the top level of my new policy I assigned my business role and the event that is being used (InteractionEnded).
4) I check business role customizing: function profile "INTENT_DRIVEN_INTERACTIONS" is assigned to "DEFAULT". Note: The intent driven profile "DEFAULT" is assigned to the service mgr. profile "IDI_DEFAULT".
Doing these changes my system now always jumps to the inbox when I click on the "END" button.
The problem seems to be that I am not able to figure out how to use the currentInboxItem as part of my rule policy.
I added a line in my rule for the currentInboxItem checking if the currentInboxItem is not-equal to "blank". This seems not to work. I guess I must rather check if currentinboxitem is "bound" or against a field of the item. But I don't see how to do this.
Has s.b. any idea how to get this work? Or on other ways how to get this work?
Does s.b. know how to debug when a rule is checked by the system?
Thanks & Regards,
ThorstenHi Paul,
I defined the execute method in the following way.
You can see below that lo_activity is defined in the following way:
lo_activity TYPE REF TO if_bol_bo_property_access,
CURRENTINBOXITEM' is a standard object of the global data container which gets filled when an inbox item gets selected.
'ZCURRENTINBOXITM' is the repository object of the context ICRule and I fill the parameter for the process type.
Hope this helps.
Regards, Thorsten
METHOD if_crm_smf_service~execute.
fill attributes for currentinboxitem
see implementation example in class CL_CRM_IC_F_G_TRANS
DECLARATION -
DATA :
lv_string TYPE string,
lo_bdc TYPE REF TO if_crm_ui_data_context,
lo_activity TYPE REF TO if_bol_bo_property_access,
lo_fb TYPE REF TO cl_crm_smf_fact_base,
lv_entity TYPE REF TO cl_crm_bol_entity,
lv_rel TYPE REF TO cl_crm_bol_entity,
lv_factbase TYPE string.
FIELD-SYMBOLS :
<lv_date> TYPE sydatum,
<lv_time> TYPE sytime.
CODE -
TRY.
CALL METHOD super->if_crm_smf_service~execute
EXPORTING
session = session
service_properties = service_properties.
CATCH cx_root.
ENDTRY.
lo_bdc = cl_crm_ui_data_context_srv=>get_instance( ).
get entity activities
lo_activity ?= lo_bdc->get_entity( 'CURRENTINBOXITEM' ).
prepare the nodes
lo_fb = session->get_factbase( ).
IF lo_activity IS BOUND.
TRY.
get only one order objects
lv_string = lo_activity->get_property_as_string( iv_attr_name = 'CRM_GUID' ).
lv_entity ?= lo_activity.
lv_rel = lv_entity->get_related_entity('BTOrderHeader').
CATCH cx_root.
ENDTRY.
IF lv_rel IS BOUND.
TRY.
CLEAR lv_string.
curr.inboxitem filled
lv_string = 'X'.
CALL METHOD me->set_fb_attr_by_id
EXPORTING
id = 'ZCURRENTINBOXITM'
value = lv_string.
process_type
CLEAR lv_string.
lv_string = lv_rel->get_property_as_string( iv_attr_name = 'PROCESS_TYPE' ).
create a child node and add the value
lo_node1->set_attribute( attrib_name = 'BT_PROCESS_TYPE'
attrib_value = lv_string ).
CALL METHOD me->set_fb_attr_by_id
EXPORTING
id = 'ZCII_PROC_TYPE'
value = lv_string.
CATCH cx_root.
ENDTRY.
ELSE.
CLEAR lv_string.
not filled
lv_string = space.
CALL METHOD me->set_fb_attr_by_id
EXPORTING
id = 'ZCURRENTINBOXITM'
value = lv_string.
ENDIF.
get next BTOrderHeader
lo_activity = bdc->activities->if_bol_bo_col~get_next( ).
ENDIF.
lv_factbase = lo_fb->get_factbase_as_string( ).
ENDMETHOD. -
(Continuity) Caller Id name not appearing in mac
Hi every body
I face problem with some calls, the Caller Id name sometimes not appearing in incoming call for mac (Continuity) only numbers.
Please note that all the calls i get it has name in my contacts and it's already in iPhone and mac contacts app.
Any ideas how to fix it.Thanks for all your reply,
I found that we have Avaya Definity System with S8700 & found configuration documentation :
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/pbx/interop/notes/410364av.pdf
What I found that Avaya side is configured for Qsig ; Basic Supplementary Services is not enabled (System Parameter & Customer options) .
I also found this document ( http://www.svcc.cc.il.us/UserMenu/employees/telephone/images/buttons.pdf Page 362) about Supplementary Service protocol value which most likely change which Qsig Avaya side use.
Once again thank you all , will let you know if things work . -
Conversion error in JCo outbound call
hi all
i am working on making OUTBOUND call from SAP using JCo,
our SAP system is Unicoded and i am refering Jco example 5 from JCo2.1.8 demo.
i am able to make connection successfully , but while running test ABAP program for testing connection, i am getting Short dump error headed :
Conversion error "ECHOTEXT" from character set 4103 to character set 4102
A conversion error occurred during the execution of a Remote Function
*Call. This happened either when the data was received * or when it was
*sent. The latter case can only occur if the data is sent from * a Unicode
system to a non-Unicode system.
The current program had to be terminated because of an
error when installing the R/3 System.
my ABAP prg. is this :
REPORT z_jco_test.
PARAMETERS: requtext LIKE sy-lisel.
*ECHOTEXT LIKE SY-LISEL
**" RESPTEXT LIKE SY-LISEL
DATA: echotext type SYLISEL,
resptext LIKE sy-lisel,
rfctest TYPE TABLE OF rfctest,
wa_rfctest TYPE rfctest.
wa_rfctest-rfcdata1 = requtext.
wa_rfctest-rfcdata2 = 'Hello World'.
APPEND wa_rfctest TO rfctest.
CALL FUNCTION 'RFC_PING'
DESTINATION 'JCO'.
*CALL FUNCTION 'STFC_CONNECTION'
* EXPORTING
* requtext =
** IMPORTING
** ECHOTEXT =
** RESPTEXT =
CALL FUNCTION 'STFC_CONNECTION'
DESTINATION 'JCO'
EXPORTING
requtext = requtext
IMPORTING
echotext = echotext
resptext = resptext
TABLES
rfctest = rfctest.
WRITE: 'Echo Text: ', echotext.
WRITE: 'Response Text: ', resptext.
LOOP AT rfctest INTO wa_rfctest.
WRITE: / 'rfcdata1: ', wa_rfctest-rfcdata1.
WRITE: / 'rfcdata2: ', wa_rfctest-rfcdata2.
ENDLOOP.
i have made RFC connection unicode yes.
please guide meHelo!
We have a similar problem here. Could you please explain your solution in a bit more detail ?
Thanks in advance,
Markus -
Outbound call not working for Common Area Phone
Hi,
I have configured Common area phone and i am not able to make the outbound call. I have grant the conference policy. I have grant the voice policy.
client policy is default
How to troubleshoot the issue with outbound calls?
INBOUND call is functioning fine
Thanks
jitenderYou can change the dial plan with my tool :)
https://gallery.technet.microsoft.com/Lync-2013-Common-Area-57bc4ff1
But good call, normalization rules may be off due to default global dial plan.
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications
This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs. -
Outbound Call in Cisco Finesse
Anybody please help what all configuration to be done in Finesse to do the outbound calls.
I am using Cisco UCCX 10.0, in which i have both the option for Cisco CAD and Finesse. When i run a outbound campaign its coming as an incoming call in Finesse.
When i answer the call, its showing the error "Unable to communicate with Enterprise Server. Outbound option not available".
Please help how to enable the outbound controls in FinesseThis is ineed the case, from RN (somewhat cryptic :-) ):
Cisco Finesse
Cisco Finesse is the next generation browser-based agent and supervisor desktop for Unified CCX. Finesse is an alternative to Cisco Agent Desktop, Cisco Supervisor Desktop, and Cisco Desktop Administrator. Finesse is available with Enhanced and Premium license packages and provides typical inbound voice contact center functionality. It supports Unified Communications Manager-based silent monitoring and workflow-based recording with MediaSense and Work Force Optimization (WFO).
Chris -
Adding *67 to all outbound calls NOT in my contacts?
Sorry if this is a redundant question, I could not find an answer on a search...is there any way to either block caller ID, or automatically insert a *67 when calling numbers NOT in my contacts? Thanks.
I know I can do that, but would like to have some way to do it automatically. I use my phone for both personal and work, and customers I call are not in my contacts, so it would be extremely helpful to have a way that when I call a number that's not in my contacts, that it automatically inserts a *67, or some other way to blcok outbound caller ID. I think the blackbery can do it, I was just curious if the iPhone can...but it sounds like it's not an option.
-
Lync 2011 for Mac with iMessage on OSX inbound and outbound calls fail
Running Office 14.4.9 updates and many before that. If you activate or deactivate an iMessage account using the built-in Messages application, Lync can no longer make or receive calls. Calls received will automatically go to voicemail.
Outbound calls just ring and ring. The phone setting in Lync were all three set to Lync as well.
The only way I've found them to work is to following the first instructions at the following link:
https://support.microsoft.com/en-us/kb/kbview/2691870?wa=wsignin1.0
Log on to your computer by using administrative credentials.
Exit Lync if it's running.
Drag the Lync application to the Trash.
To remove your existing Lync preferences, delete the following files:
Users/username/Library/Preferences/com.microsoft.Lync.plist
Users/username/Library/Preferences/ByHost/MicrosoftLyncRegistrationDB.xxxx.plist
Users/username/Library/Logs/Microsoft-Lync-x.log
If you activate or deactivate an iMessage account steps 1-4 must be performed, otherwise, phone calls fail. One additional annoyance I've noticed is that Lync tends to hand with a red X on the icon in the dock when this happens as well. You must
use the terminal and do 'pkill Lync' in order to close it.Hi MSGuest,
Can you install the latest update and then check again ?
https://www.microsoft.com/en-us/download/details.aspx?id=36517
Best regards,
Eric
Please remember to mark the replies as answers if they help, and unmark the answers if they provide no help. If you have feedback for TechNet Support, contact [email protected] -
Response Group to PSTN Outbound Caller ID
Hi Guys,
I'm trying to implement an after hours paging service on our lync system using a response group (with overflow set to 0), but the problem I'm currently facing is that the outbound calling number ID is set to the number of the caller, which I'm not legally
allowed to present here in Australia. I could in theory set the number to be private, but that will cause people to screen the calls.
Is there any way to get the RGS application to set the call history in the same way that simultaneous ringing does, even perhaps on some server side-scripting?
This is the SIP info I get in a simultaneous ring, where xxx is the number they called, and yyy being the number they called from.
HISTORY-INFO: <sip:+xxx@frontend;user=phone>;index=1,<sip:+yyy@frontend;user=phone>;index=1.1
Any input would be very appreciated :)I'm curious what happens if you try this:
You'd create a new voice policy, PSTN usage and route. On the route, you mask the caller ID to be your business's primary number.
Then assign that policy to a user account that you've created just for this purpose. That user account should forward it's calls to the number the response group should call.
Then, with the overflow still set to 0, instead of a calling the number, you call that forwarding user account with the caller id mask.
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications
This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.
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