IPIVR for outbound call?
IPIVR seems be designed for handling inbound call. Is it possible that ipivr can handle scheduled outbound calls?
Or is there any products (cisco or others) for this functions?
thanks first
You can make outbound calls from a script; however, they must be triggered by an external event. This is typically done by an HTTP trigger to the script which then makes an outbound call leg based on parameters provided through the HTTP trigger. I.E. your application server issues the HTTP trigger to IPIVR to make a call when the appropriate event occurs.
If you are looking for an outbound dialer, this is included in CCX Premium for the quantity of agents you purchase.
Similar Messages
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UCCX HR Report for Outbound Calls
Hi,
We are using UCCX 9.0.
In Historical report we able to get the report for outbound call also.
The issue for the Outbound is calls which are not answered by the remote party There is also a call duration though the CUCM CDR shows zero seconds and the Agent Call Records in CAD shows the call and unanswered.
Because of this we are not able to identify the matured calls & non matured calls for the outgoing.
Is there anyway to capture unanswered outbound calls from Historical reports
Thanks,
PaulI must say that I had similar problems with reports for outbound calls on our CC system...after some writings with TAC we concluded that reports for outbound calls in CC system are much more less informational versus inbound calls. So what I want to say is that probably your reporting is working fine and there is just no info for outbound calls which can be utilized and be usefull for you in your reports...
BR,
Dragan -
Another layout for outbound call in ICM 7.5
Hello everyone,
If someone knows whether it is possible to make another layout which I have to present in CAD for outbound calls.
I have already used default layout and need another one for outbound calls.
Our contact cener is based on ICM 7.5 with IVR and CUCM.
Thanks.Use the default layout for outbound calls, and create a separate custom layout for all other types of calls.
-
DTMF doesn't work for outbound call
We currently using UC520 and having strange problem. When we call through SIP for outbound call DTMF tone is not transmitting. and if I press multiple numbers with longer duration, It makes continuous tone on the other hand.
My current dialpeer uses DTMF relay RTP- NTE. I will attach sh run and debug ccsip message file. please help me this is being quite critical.Seems like your SIP Trunk Service provider is not getting the RFC2833 packets or not interpreting them correctly.
I see on the log that you pressed 1, 2 & 3 and the UC520 sends all these digits out (each digit will have 7 packets - this post explains how to read this debug - https://www.myciscocommunity.com/message/14045#14045). So I know its not the UC520 - below is a snip for digit 1
Sep 10 16:26:26.114: s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B86 timestamp 0xBCB7190A
Sep 10 16:26:26.114: Pt:101 Evt:1 Pkt:04 00 00 >>
Sep 10 16:26:26.114: s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B87 timestamp 0xBCB7190A
Sep 10 16:26:26.114: Pt:101 Evt:1 Pkt:04 00 00 >>
Sep 10 16:26:26.114: s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B88 timestamp 0xBCB7190A
Sep 10 16:26:26.114: Pt:101 Evt:1 Pkt:04 00 00 >>
Sep 10 16:26:26.114: s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B89 timestamp 0xBCB7190A
Sep 10 16:26:26.114: Pt:101 Evt:1 Pkt:04 01 90 >>
Sep 10 16:26:26.114: s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B8A timestamp 0xBCB7190A
Sep 10 16:26:26.114: Pt:101 Evt:1 Pkt:84 03 20 >>
Sep 10 16:26:26.114: s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B8B timestamp 0xBCB7190A
Sep 10 16:26:26.114: Pt:101 Evt:1 Pkt:84 03 20 >>
Sep 10 16:26:26.114: s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B8C timestamp 0xBCB7190A
Sep 10 16:26:26.114: Pt:101 Evt:1 Pkt:84 03 20 >>
In terms of next steps:
- Is there any router or NAT or firewall device between the UC520 WAN interface and the SIP Trunk service provider? If so maybe good to check if you can bypass this device or if this device blocks RFC2833 packets
- Ask the SIP Trunk SP what they see on their end for the RFC2833 packets - you can send them a wireshark capture off the UC500 WAN port if that is more convincing than a debug log. Who is the SP by the way? -
Use Cisco CUCM for outbound "call me at" feature on Lync meetings
I'm trying to find a step by step to enable users (non enterprise voice users) to use the dial me at feature in Lync conference meetings. I only want the user to have the ability to tell Lync to dial a number to place that number into the conference call,
the feature is easy to enable but i can't get the routing right between CUCM and Lync. I've looked all around the net but I can't seem to find anything that matches what i'm trying to do, other docs cover enterprise voice and that's out of my scope. Any assistance
here would be nice. ThanksHi,
In Lync Server 2010, it is not supported with “call me at” function for non-Enterprise Voice users.
However, Lync Server 2013 now allows participants that are not Enterprise Voice enabled to initiate dial-out calls from a meeting conference, called “Dial-out Conferencing for non-Enterprise Voice users”.
This can be configured by setting the Conferencing policy to allow this feature (Set-CSConferencingPolicy –AllowNonEnterpriseVoiceUsersToDialOut:$true). After enabling this, then assign a voice policy to the users who need the function.
Best Regards,
Eason Huang
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]
Eason Huang
TechNet Community Support -
Play a Prompt for outbound call
Hi,
I have an application requirement as stated below:
1) An external number is dialed by Cisco IPT user
2)Once the call is connected the user is given an option to play a pre configured prompt say "This is a call from MyTechnologies"
3)Once the IVR plays the prompt it should automatically drop out of the conference.
4)The two parties will continue talking to each other afterwards.
How can I achieve the above using CUAE.
Regards
PranjolI'm also interested in knowing how to play a prompt after a call has started and then the two parties keep talking:
Something like:
Dial the number in the IP phone.
Play the prompt: "This call is being recorded"
Then talk: "Hello, how can I help... bla bla"
Thanks in Advance, -
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 -
Not all my outbound calls g.711?
hi
have added a new region device pool etc to callmanager..have noticed on some of my new phones that some of the outbound calls show up as g.711 which is correct and other as g.729..not sure why this is as new gateway is in same device pool as new phones and callmanager is set to use g.711 within this region and g.729 to other regions..feel its something on the gateway or am i missing somethingSometimes some calls don't match any configured dial-peers and use default dial-peer. Default dial peer always uses g729.
Also, it is possible that remote phone configured for g729 region uses your gateway for outbound calls.
useful commands:
show voice call status - shows which dial-peers are matching.
show call active voice id - shows complete information about the call -
Outbound call failure in TelePresence Server
Setup has CUCM - Conductor - TelePresence Server (virtual). Plan is to use the same setup for scheduled conferences by including TMS. I have done all configuration as per the latest Conductor with TMS deployment guide.
While testing calls, I could see that the conference is getting created in the TelePresence server and the TelePresence server is trying to make a outbound call to the endpoint SIP address (extn@CUCMIP). But the calls are not getting completed.
If I configure TLS in the SIP settings of TS for outbound calls, then I am getting the below in the TS logs.
698
13:33:51.845
APP
Info
conference "Scheduled_Conference_zzzz": deleted via API (no participants)
697
13:29:41.040
APP
Info
call 14: tearing down call to "[email protected]" - destroy at far end request; networkError
696
13:29:41.040
CMGR
Info
call 14: disconnecting, "[email protected]" - network error
695
13:29:41.039
SIP
Error
call 14: Ending call due to network error during INVITE transaction
694
13:29:40.539
APP
Info
call 13: tearing down call to "[email protected]" - destroy at far end request; networkError
693
13:29:40.539
CMGR
Info
call 13: disconnecting, "[email protected]" - network error
692
13:29:40.539
SIP
Error
call 13: Ending call due to network error during INVITE transaction
691
13:29:08.544
APP
Info
call 14: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
690
13:29:08.437
APP
Info
call 13: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
689
13:29:07.765
APP
Info
conference "Scheduled_Conference_zzzz" created
If I use TCP in the SIP settings, I am getting the below in the TS logs.
688
13:03:51.822
APP
Info
conference "Scheduled_Conference_zzzz": deleted via API (no participants)
687
13:01:28.141
NTP
Info
time is Tue Apr 28 13:01:28 2015
686
13:00:32.121
APP
Info
call 12: tearing down call to "[email protected]" - destroy at far end request; unavailable
685
13:00:32.121
CMGR
Info
call 12: disconnecting, "[email protected]" - service unavailable
684
13:00:32.109
APP
Info
call 12: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
683
13:00:32.009
APP
Info
call 11: tearing down call to "[email protected]" - destroy at far end request; unavailable
682
13:00:32.009
CMGR
Info
call 11: disconnecting, "[email protected]" - service unavailable
681
13:00:31.996
APP
Info
call 11: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
680
13:00:01.955
APP
Info
call 10: tearing down call to "[email protected]" - destroy at far end request; unavailable
679
13:00:01.954
CMGR
Info
call 10: disconnecting, "[email protected]" - service unavailable
678
13:00:01.936
APP
Info
call 10: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
Some of the questions which are not answered in the guide are :
Is a new SIP trunk required from CUCM to Conductor. If yes, what is the destination IP for this trunk. Is this the primary conductor IP address. For adhoc & rendezvous conferences, there are seperate SIP trunks created and destination IP is the additional IP address configured.
Any other configuration required in any of the other applications.
Thanks.Setup has CUCM - Conductor - TelePresence Server (virtual). Plan is to use the same setup for scheduled conferences by including TMS. I have done all configuration as per the latest Conductor with TMS deployment guide.
While testing calls, I could see that the conference is getting created in the TelePresence server and the TelePresence server is trying to make a outbound call to the endpoint SIP address (extn@CUCMIP). But the calls are not getting completed.
If I configure TLS in the SIP settings of TS for outbound calls, then I am getting the below in the TS logs.
698
13:33:51.845
APP
Info
conference "Scheduled_Conference_zzzz": deleted via API (no participants)
697
13:29:41.040
APP
Info
call 14: tearing down call to "[email protected]" - destroy at far end request; networkError
696
13:29:41.040
CMGR
Info
call 14: disconnecting, "[email protected]" - network error
695
13:29:41.039
SIP
Error
call 14: Ending call due to network error during INVITE transaction
694
13:29:40.539
APP
Info
call 13: tearing down call to "[email protected]" - destroy at far end request; networkError
693
13:29:40.539
CMGR
Info
call 13: disconnecting, "[email protected]" - network error
692
13:29:40.539
SIP
Error
call 13: Ending call due to network error during INVITE transaction
691
13:29:08.544
APP
Info
call 14: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
690
13:29:08.437
APP
Info
call 13: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
689
13:29:07.765
APP
Info
conference "Scheduled_Conference_zzzz" created
If I use TCP in the SIP settings, I am getting the below in the TS logs.
688
13:03:51.822
APP
Info
conference "Scheduled_Conference_zzzz": deleted via API (no participants)
687
13:01:28.141
NTP
Info
time is Tue Apr 28 13:01:28 2015
686
13:00:32.121
APP
Info
call 12: tearing down call to "[email protected]" - destroy at far end request; unavailable
685
13:00:32.121
CMGR
Info
call 12: disconnecting, "[email protected]" - service unavailable
684
13:00:32.109
APP
Info
call 12: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
683
13:00:32.009
APP
Info
call 11: tearing down call to "[email protected]" - destroy at far end request; unavailable
682
13:00:32.009
CMGR
Info
call 11: disconnecting, "[email protected]" - service unavailable
681
13:00:31.996
APP
Info
call 11: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
680
13:00:01.955
APP
Info
call 10: tearing down call to "[email protected]" - destroy at far end request; unavailable
679
13:00:01.954
CMGR
Info
call 10: disconnecting, "[email protected]" - service unavailable
678
13:00:01.936
APP
Info
call 10: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
Some of the questions which are not answered in the guide are :
Is a new SIP trunk required from CUCM to Conductor. If yes, what is the destination IP for this trunk. Is this the primary conductor IP address. For adhoc & rendezvous conferences, there are seperate SIP trunks created and destination IP is the additional IP address configured.
Any other configuration required in any of the other applications.
Thanks. -
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. -
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] -
Wrap-up time for manual outbound calls (UCCX)
Is it possible to configure the wrap-up time for the manual outbound calls in UCCX? I think, this option only exists in CSQ, which is of course meant for the inbound calls. Any thoughts or any workaround to make this work?
Requirement- Once a manual outbound call is hung up, agent's state should be switched to WORK READY as per the wrap-up timer setting.
Thanks.Dear experts,
I look forward to hear from you if you have anything to offer. Wrap-up TIME to be setup for manual outbound calls in UCCX.
Thanks,
Piyush -
Phone won't make outbound calls. can't call verizon support. this after being on hold for 17 minutes in an attempt to fix voicemail notification. this is a nightmare and Verizon is unreachable. Any way to get Verizon wireless to fix my service when they can't be called (because their phone service doesn't work)?
Well 1, I guess it's not that important to you, 2, customer support generally closes at 11pm in the time zone you are in, 3, toll free and local calls are free in hotels, 4 there is limited support after hours.
-
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. -
To create Trunk Group with differnt T1 PRI's groups for our outbound calls
Hi All,
I would like to request all of you that I have requiremnt that we have to create "Trunk Group" of diffenent T1 controller PRI's for outbonds call, mean I need to create a trunk group with differnt T1 PRI's groups for our outbond calls, but my problem is that I don't know and no idea that how I will do this , i am also try to find some cisco doc for this still I did not find these info, so I request all of you that I will be thank full to all of you if you can help me out for my this problems.
Thanks
RizwanHere is a good write up, there are many others:
http://www.markholloway.com/blog/?p=452
HTH,
Chris
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