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.

Similar Messages

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,
Paul

I 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

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?

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.

Another solution for the "caller can't hear me" pr...

Ran into a problem with my cell phone the other day... I could hear the caller, but he could not hear me. Also, the Voice Recorder would not hear me either. All of which suggested the microphone was not working.
I rummaged the web, tried the various solutions I found (loudspeaker on and off; phone on and off; reset device--both soft and hard; removed battery and SIM, too), and even called AT&T customer service, AT&T GoPhone service, and Nokia (they all suggested the reset steps). No luck whatsoever. The general consensus was that the phone was a goner and I needed to get a new one. I was actually told that 2-2 1/2 years is about what should be expected for a cell phone's longevity!
The problem began when I inadvertently hit the camera key in the middle of the call. So, I even tried Mute on and off in the camera's video recorder
In any event, I finally tumbled to a solution when I noticed the inside light of the cell phone shining through its side, at the bottom. If I pressed down or squeezed the case, the microphone started working again! Turns out that the screws inside that hold the innards to the front of the phone had gotten quite loose. They are accessible when you removed the battery cover. One of mine actually dropped out when I took the battery cover off. Once I screwed them back in tightly, the microphone was back working like a charm.
Note: The screws have some of sort of mini-Allen head type crevice for their tops (a torx of some sort?), so you'll need to be careful with what you use to screw them tight.
In any event, I'm posting this here because this was not covered anywhere else on the web and the tech people at the various companies didn't know about it either.
By the way, before you ever do a hard reset, make sure that you've backed up your stuff first and, whether soft or hard reset, make sure that you write down which settings you've tweaked-- main display, icons, custom keys, etc.--because you'll have to redo them later. That'll save you lots of time and remembering!
On the Mac, for backing up, you can use iSync even with "unsupported" devices--search around there are tips on how to dig into iSync's contents and add your device. Again, though, I would try the other solutions first before resorting to that.

iphone hardware troubleshooting http://support.apple.com/kb/TS2802

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. Thanks

Hi,
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

Billed by another provider for collect calls accepted from my Verizon landline

We accepted two brief (under 3 Minutes) emergency collect phone calls from a family member to our Verizon landline, We were billed $39.36 for them by a "provider" called ILD. I called Verizon and a representative said there is nothing they can do and to call ILD. I called ILD and they offered an $11.00 "discount" which I told them was unacceptable. They told me to call Verizon back, which I am not going to do right now because each phone call involved substantial time to get through automated message handling systems and finally get an actual person to talk with.
I was frustrated with Verizon to begin with because of their outrageously high charges, constant increases, complicated bills and terrible customer service. Tomorrow, I will do one or all of the things below:
(1) Contact consumer affairs and government agencies who may be able to help me.
(2) Cancel my current Verizon phone and internet service completely.
(3) Never deal with Verizon again for any type of services and advise friends/relatives to do the same.
I made a partial payment of $100 on my bill. This covers all Verizon services and $10 towards ILD charges, which is all I am willing to pay.

Your issue has been escalated to a Verizon agent. Please go to your profile page for the forum, and look in the middle, right at the top where you will find an area titled "My Support Cases". You can reach your profile page by clicking on your name beside your post, or at the top left of this page underneath the title of the board.
Under “My Support Cases” you will find a link to the private board where you and the agent may exchange information. This should be checked on a frequent basis as the agent may be waiting for information from you before they can proceed with any actions. Please keep all correspondence regarding your issue in the private support portal.

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
Pranjol

I'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 something

Sometimes 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

Can we call two layouts for a single script?

Can we call two layouts for a single script?
Where dow e assign print program to a script?
in NACE transaction is it possible.......
Thanks in advance.
Regards.
Abhilash.

Hi Abhilash.
Greetings for the day.
U can use one script in diffrent programs.
Yea u can do it in NACE transaction.
procedure would b as foloows:-
1.NACE
2.Select the apllication
3.Output types (press tab).
4.again selet the output type which u require
5.then press "procedure routines"
here u can give the program name and fom name.
plz reward if found helpful.
regards
prashant tiwari

Another layout for outbound call in ICM 7.5

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