Call Handler Transfer to Meet Me Conference

I am struggling a bit with setting up a call handler in Cisco Unity 7.0.
We have a DID direct to the Unity administrator. They want the ability to dial the MeetMe conference numbers (799X) and dial into them from an outside line without having to call the reception and ask to be transferred.
I have setup a Call Handler for Ext. 7999, set the greeting to none, set the Call Transfer to release to phone at Ext. 7999 and 'Release to switch'. I receive a message stating, "That extension can not be dialed". What am I missing?
Regards,
Jason

Are you allowing that in your restriction tables?

Similar Messages

Call Handler Transfer Issue

I have a Unity 4.0 system that is exhibiting the following behavior.
Inbound calls go to voice mail. Recorded prompt notifies caller that they can press 1 to transfer to cell number.
Caller input within the Unity subscribers account is programmed to "Send caller to attempt transfer for [a specific Call handler that is set to]" "Yes, ring a subscriber at this extension: [users cell number]"
When the caller calls the users DID, the call flow completes correctly without any issues.
When the caller calls any of our main lines, we have several published company phone numbers, and then either enters the end users extension, which is allowed via a recorded attendant, or chooses the operator and is then manually transfered to the end users extension, the caller, upon pressing 1 to transfer to the cell phone experiences one of the 4 following behaviors.
1. The call transfers to the cell phone but the call terminates before the end user succesfully answers the call.
2. The caller hears the prompt "your call can not be completed as dialed, [Wireless Cellular Carrier Name]".
3. Example Administrator is not available.
4. A Voice Mail tone prompt without any preceding recording.
I have over a dozen out of 200 + accounts programmed for this feature. All accounts are utilizing the same PRIs for inbound and outbound calls. All accounts use the same gateways, interfaces, Calling Search Space, Partition, Device Pool, Route Group, Route List and Route Pattern.
Of these accounts that utilize this feature programming, 3, including a test phone that I just set up patterned after one account that is exhibiting this unwanted behavior, function correctly no matter where the call comes from.
Originally, all accounts exhibited this unwanted behavior but after modifying all associated call handlers with the following, the three afore mentioned accounts function correctly.
Changes to call handler: Changed Transfer Type from "Release to Switch" to "Supervise Transfer" and "Rings to Wait" from 2 to 3.
Any suggestions and advice are greatly appreciated.

Here is an update to this issue. Through additional testing I've determined the following.
Unless the transfered call is answered by the cell phone immediately, the call transfer fails and the call is instead transfered to the message recipient set within the call handler. By default, this is Example Administrator. I can change the owner of and message recipient of the call handler and the call does go into that subscriber's voice mail without any greeting.
Again, this behavior only occurs when the originating call is transfered from one call handler to the transfer to cell phone call handler. If the originating call is to the users DID, the call transfers without any issues.

How can I change the default message after System Call Handler transfer error?

Hi,
I'm using Unity Connection 9.1 and System Call Handlers to allow to join internal extensions (not reachable from the PSTN).
I succeeded but I would like to change the default message played when the extension dialed is not known. I tried with Error greetings but nothing changed.
Thanks for your help.
Regards,
Julien

Just tried this on my 9.1 and 10.0(1) servers and it worked fine.
I went to the opening greeting call handler, selected the "error" greeting, set the greeting to play my personal greeting (it defaults to the system generated greeting), recording "hey, you dialed the wrong number" and then called in and during the opening greeting I dialed "88" - I heard "hey, you dialed the wrong number" and then it repeated the opening greeting recording as it should.
the default action after playing your custom greeting in this case is to "restart greeting" - which is why the opening greeting plays again.
Not sure what you did in your testing with the error greeting - check to make sure you have it set to play your personal greeting instead of the default system greeting - since you can't disable the error greeting there's not a lot of other items I can think of to check.

Unity 7.0(2) cannot transfer from Call Handler to Non Sub Extn

Hi,
Got Unity 7 with Exchange onbox. Need to setup a Call Handler that will forward calls to another extensionon the CUCM that has no VM. I created 2 call handlers:
CH1 DN:1234 (same as CUCM DN)
Transfer incoming calls to a phone=No,send directly to Greeting
Greeting=Send caller to Call Handler CH2
CH2
Transfer incoming call to a phone=Yes. Extn 5678.
Problem is I keep getting "Sorry Example Administrator is not available, please leave a message".
Any ideas?
Rich

The call flow needs to be:
CH1 either via a caller input or an after greeting action
Attempt transfer to: CH2
In CH2 under call transfer settings, yes, ring subscriber at this DN
Your restriction tables need to allow what you're trying to dial.
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk

Call Handler - Direct Transfer.

Running Unity 4.04, I'm trying to setup a call handler to be able to transfer a call to another extension without that extension having voicemail assign to it, like a Hunt Group. I've checked the yes, ring a subscriber at this extension, but Eample Admin kicks in and the call never transfers. I've changed Profile; Active Schedule to all day too. Thanks for Any Help.
Jim

I suspect you're fiddling with the wrong greeting - this absolutely does work and is a very common techique.
As a test, set the ALTERNATE greeting for the handler you're using to sling shot the call as active, set it's source to blank and set it's after greeting action to attempt transfer for the call handler you want to ring the phone. On THAT call handler, set the ALTERNATE transfer rule to active and have it set the phone to ring.
Using the alternate greeting and the alternate transfer rule like this take the most common issue here out of the equation since they over ride all other rules without question.
This absolutely works and has always worked so if you continue to get the greeting after doing this you're liking having the call routed somewhere unexepected and you'll need to open up the call viewer and the port status montior app from the tools depot and see how the call is being processed.

Transfer to Call Handler

I have setup a new Callhandler for a Department. It works fine, never an issue Now the Department wants to forward phones to this call handler when people are away from the phones.
I cannot get it to work when you do a "CFWDALL" on the phone, you never get anything but that phones voicemail.
If the end user call forwards their phones to 12489 (call handler) you only here there voicemail never the call tree. if the phone does not not have a voicemail box then you get "Please enter the extension . . ." it never goes to the call tree.
Is there a way to get around this? I have tried creating new Voicemail Ports but that does not help.
I watch RTMT and the call shows only the original phone number never the call tree number.
I have tried creating a Translation patter 12488 and have it trnslate to 12489 but this does not work, it shows the same thing on RTMT
How do I forward a phone call to a "Call Handler"?
Thanks

Rob,
While that may work, I think that there would be an unintended side affect. To illustrate. If you had a scenario like this:
Phone_A calls Phone_B which forwards to Phone_C which forwards to voice mail.
If both Phone_B and Phone_C have voice mailboxes I would suspect that the users would want CUC to send the caller (Phone_A) to the mailbox greeting for Phone_B. If the parameter "Use Last (Rather than First) Redirecting Number" is enabled then the caller (Phone_A) would be sent to the mailbox greeting for Phone_C (rather than Phone_B). Based on my experience, users prefer to have callers integrate to the mailbox of the original redirecting party in the majority of use case scenarios.
Neal,
While this setting fixes this one issue it may introduce other issues. Certainly Hailey, Rob, Java are all capable of correcting me if I am wrong on this one.
Assuming I am correct, then I think I would go with option 4 and then add the appropriate Forwarded Routing Rule under Call Management>Call Routing. You will want to place the rule in front of "Attempt Forward" and modify the information element fields accordingly.
HTH.
Regards,
Bill

Meet Now Conference Invite - UCMA

Hi all,
Back again, I'm currently looking into getting into the signalling path of Meet Now conferences.
Currently I can get the initiator successfully into the UCMA application and B2B the call out to the original destination. This works fine and the client can then add new participants
The invite which is passed to UCMA via MSPL is the Invite which contains the SDP.
Upon adding a participant, the participant client rings, and upon answer the participant is ejected from the conference. I never the see the Invite hit the UCMA application, although I do see the Invite in the SessionReceivedEventArgs.
The UCMA application rejects the invite with a 403 Forbidden and the reason is "Remote participant mismatch".
Can anyone shed any light on this? I can't see any reason for the rejection.
Thanks
Michael.
TL_INFO(TF_PROTOCOL) [1]270C.22E4::03/25/2015-14:59:22.438.005d93c1 (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[1979629883] $$begin_record
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TL_INFO(TF_DIAG) [1]270C.22E4::03/25/2015-14:59:22.443.005d9805 (SIPStack,SIPAdminLog::WriteDiagnosticEvent:SIPAdminLog.cpp(802))[1979629883] $$begin_record
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TL_INFO(TF_PROTOCOL) [1]270C.22E4::03/25/2015-14:59:22.443.005d980a (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[1979629883] $$begin_record
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TL_INFO(TF_PROTOCOL) [1]270C.22E4::03/25/2015-14:59:22.455.005d9d7c (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[2374122616] $$begin_record
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TL_INFO(TF_PROTOCOL) [1]270C.22E4::03/25/2015-14:59:22.504.005da228 (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[2374122616] $$begin_record
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Peer: vm-dev-lyncdev.twdev01.com:5061
Message-Type: response
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TO: <sip:[email protected];gruu;opaque=app:conf:audio-video:id:SD4DPLJ6>;epid=95364369E6;tag=8c22af388d
CALL-ID: c6ce41de373f47d8bc3c91231fdea0f6
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TL_INFO(TF_DIAG) [1]270C.22E4::03/25/2015-14:59:22.504.005da363 (SIPStack,SIPAdminLog::WriteDiagnosticEvent:SIPAdminLog.cpp(802))[3077166619] $$begin_record
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Text: Routed a request on behalf of an application
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Data: application="http://www.twdev01.com/Archiver"
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TL_INFO(TF_PROTOCOL) [1]270C.22E4::03/25/2015-14:59:22.504.005da368 (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[3077166619] $$begin_record
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Instance-Id: 19523
Direction: outgoing;source="local"
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Message-Type: request
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ms-diagnostics-public: 5012;reason="ACK is being generated on receipt of a failure final response for an INVITE forked by application";AppUri="http%3A%2F%2Fwww.twdev01.com%2FArchiver"
$$end_record
TL_INFO(TF_DIAG) [0]270C.22E4::03/25/2015-14:59:22.505.005da641 (SIPStack,SIPAdminLog::WriteDiagnosticEvent:SIPAdminLog.cpp(802))[2374122616] $$begin_record
Severity: information
Text: Response successfully routed
SIP-Start-Line: SIP/2.0 403 Forbidden
SIP-Call-ID: c6ce41de373f47d8bc3c91231fdea0f6
SIP-CSeq: 1 INVITE
Peer: 10.1.5.81:58485
Data: destination="[email protected]"
$$end_record
TL_INFO(TF_PROTOCOL) [0]270C.22E4::03/25/2015-14:59:22.506.005da646 (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[2374122616] $$begin_record
Trace-Correlation-Id: 2374122616
Instance-Id: 19522
Direction: outgoing
Peer: 10.1.5.81:58485
Message-Type: response
Start-Line: SIP/2.0 403 Forbidden
FROM: "Lync4"<sip:[email protected]>;tag=1f9f0e6509;epid=4dada7ef28
TO: <sip:[email protected];gruu;opaque=app:conf:audio-video:id:SD4DPLJ6>;epid=95364369E6;tag=8c22af388d
CALL-ID: c6ce41de373f47d8bc3c91231fdea0f6
CSEQ: 1 INVITE
Via: SIP/2.0/TLS 10.1.5.81:58485;ms-received-port=58485;ms-received-cid=203900
CONTENT-LENGTH: 0
ms-diagnostics: 24019;Component="RTCC/5.0.0.0_Test Lync Redirector";Reason="Remote participant mismatch";Source="vm-dev-lyncdev.twdev01.com"
P-ASSERTED-IDENTITY: <sip:[email protected];gruu;opaque=app:conf:audio-video:id:SD4DPLJ6>
$$end_record
TL_INFO(TF_PROTOCOL) [1]270C.22E4::03/25/2015-14:59:22.518.005da65d (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[1979629883] $$begin_record
Trace-Correlation-Id: 1979629883
Instance-Id: 19525
Direction: incoming
Peer: 10.1.5.81:58485
Message-Type: request
Start-Line: ACK sip:[email protected];gruu;opaque=app:conf:audio-video:id:SD4DPLJ6 SIP/2.0
From: <sip:[email protected]>;tag=1f9f0e6509;epid=4dada7ef28
To: <sip:[email protected];gruu;opaque=app:conf:audio-video:id:SD4DPLJ6>;epid=95364369E6;tag=8c22af388d
Call-ID: c6ce41de373f47d8bc3c91231fdea0f6
CSeq: 1 ACK
Via: SIP/2.0/TLS 10.1.5.81:58485
Max-Forwards: 70
Content-Length: 0
$$end_record

Hi,
Agree with Holger, I check on my Lync Windows Store app and can't find the meeting URL as well.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Unity call handler problem

Hello,
If somebody can help me with Unity configuration, I will be appritiated.
I`ve got a number 555 with system call handler configured on Unity and all call gets forwarded to VM. When I am calling to 555 from my phone everything works fine. But when I try to forward all calls from another extention, let`s say from 444 to 555, I am hiting default Unity greetings for the number 555 instead of system one.
Any ideas much appritiated !
Maxim

Hi Maxim,
This is the expected behavior as Unity will see the Forwarded number
of 444 instead of 555 and doesn't know what to do with it.
So, let's say you want to route 444 to the 555 Call Handler.
You could create a Voicemail Profile in CUCM called "555 Transfer or whatever" which points to DN 555 (**Voicemail Box Mask = 555).
Apply this new profile to 444 (under DN config page VM-Profile in CUCM) and when calls route through 444 and forward to voicemail they will receive the 555 Call Handler Greeting etc.
You could also use the Forwarded Routing Rule set up in Unity to create a rule for this 444 to 555 forwarding.
Cheers!
Rob
"I don't know how, I don't know when
But you and I will meet again "
- Tom Petty

No Call Forward to Call Handler

Hi,
we have cucm 8.6 and cuc 8.6
inbound call from ITSP transferred to CUC call handler for opening greetings and by pressing '0' it transfers to operator.
i want to configure custom busy message when operator is busy with other calls.
i configured another call handler with number 1660 and forwarded the operator calls to the call handler if busy.
the busy message is playing when i dialed the number 1660 from any extension, but the calls not getting transferred when the operator is busy.
i tried with forward all calls to the callhandler and getting the message ' cannot be reachable'.
i configured busy message on opening greeting call handler itself, in the callhandler "greetings" section's "busy" option, it didn't work as the call already got transferred to cucm.so its not coming back to callhandler if the ext is busy.
please can anyone guide me to configure busy message for operator.

Hi Anas,
i tried as you said,
its transferring the call if extension is busy by skipping welcome greetings, it's not what i wanted.
i need if any inbound call comes from ITSP first the welcome message with IVR plays, if the caller press '0' then only it should transfer to operator, if the operator is busy then it should play "Operators Busy" message.

Multiple Holiday Schedules on one Call Handler

Hello, I search previously for a solution to this problem, but have been unsuccessful. I am running Cisco Unity Connection 8.5.1.
Current Setup:
Company has a system call handler named "Company AA" that serves as their autoattendant for answering incoming calls.
Company has a holiday schedule associated to "Company AA" which includes recognized holidays, and an personalized greeting recording stating they are closed for the holiday.
What needs done:
Company has scheduled staff meetings throughout the year. These need to be input into the system so that on these dates/times, a different personalized greeting recording plays stating that they are closed. This needs to be a different recording than the normal Holiday greeting (so I cant just add the dates to the already existing holiday greeting).
Is this possible with Cisco Unity Connection?
Thanks!
Derek

Hello Derek,
It is possible, however, you will need to create a different Call Handler and Schedule per Holiday, in this example I am going to use just 3 holidays (July 4th, January 1st and December 25th. The process is kind of complex, but here we go.
1st: Create the Call Handlers:
Create a CallHandler for the Company AA
Create a CallHandler for January 1st
Create a CallHandler for July 4th
Create a CallHandler for December 25th
2nd: Create the Holiday Schedules
Create a Holiday Schedule for the Company AA CallHandler and add all the holidays here (January 1st, July 4th, and December 25th).
Create a Holiday Schedule for the January 1st CallHandler and add just the July 4th, and December 25th holidays here
Create a Holiday Schedule for the July 4th CallHandler and add just the December 25th holiday here
Create a Holiday Schedule for the   December 25th CallHandler and do not add any holdays
3rd: Create the System Schedules
Create a System Schedule for the Company AA CallHandler and associate it to the Company AA Holiday Schedule (Define your working/after hours schedule here).
Create a System Schedule for the January 1st CallHandler and associate it to the January 1st Holiday Schedule (Make it 24/7)
Create a System Schedule for the July 4th CallHandler and associate it to the July 4th Holiday Schedule (Make it 24/7)
Create a System Schedule for the December 25th CallHandler and associate it to the December 25th Holiday Schedule (Make it 24/7)
4rd: Call Handler configuration
Company AA
Go to the Company AA CallHandler, and under Active Schedule select the Company AA System Schedule.
Setup the Closed and Standard Greetings/Transfer Rules as you wish.
Go to the Holiday Greeting and select Nothing and uncheck the Play the "Record Your Message at the Tone" Prompt options under Callers Hear.
Under the After Greeting section, select Call Handler >> January 1st >> Go Directly to Greetings.
Check the Greeting Enabled with No End Date and Time option and save the configuration
January 1st
Go to the January 1st CallHandler, and under Active Schedule select the January 1st System Schedule.
Go to the Standard Greeting and setup the message that you want to play during January 1st here, if you want to transfer the call, you will need to change the Standard Call Transfer Rule as well.
Go to the Holiday Greeting and select Nothing and uncheck the Play the "Record Your Message at the Tone" Prompt options under Callers Hear.
Under the After Greeting section, select Call Handler >> July 4th >> Go Directly to Greetings.
Check the Greeting Enabled with No End Date and Time option and save the configuration
July 4th
Go to the July 4th CallHandler, and under Active Schedule select the July 4th System Schedule.
Go to the Standard Greeting and setup the message that you want to play during July 4th here, if you want to transfer the call, you will need to change the Standard Call Transfer Rule as well.
Go to the Holiday Greeting and select Nothing and uncheck the Play the "Record Your Message at the Tone" Prompt options under Callers Hear.
Under the After Greeting section, select Call Handler >> December 25th >> Go Directly to Greetings.
Check the Greeting Enabled with No End Date and Time option and save the configuration
December 25th
Go to the December 25th CallHandler, and under Active Schedule select the December 25th System Schedule.
Go to the Standard Greeting and setup the message that you want to play during December 25th here, if you want to transfer the call, you will need to change the Standard Call Transfer Rule as well.
Go to the Holiday Greeting and select Nothing and uncheck the Play the "Record Your Message at the Tone" Prompt options under Callers Hear.
Notes:
     As mentioned above, this process is complex and the best way to achieve this will be to create an excel table with the settings of every CallHandler so you can check them before implementing this in production, please note that the last Call Handler Holiday Schedule is empty, as well as the Holiday Greeting of this CallHandler since it will only play the Standard Greeting (24/7) schedule.
     The second
     The third thing you need keep in mind is that if you need to change the working hours schedule, you will do it on the main CallHandler only (Company AA), since the other system schedules are used only for holiday and during the holiday those are set to work 24/7.
Explanation:
     This implementation of distinctive holidays works because when the AA CallHandler check the holiday schedule during July 4th, it will send the the call to the holiday greeting since it is marked as a holiday on its schedule, the holiday greeting of the AA will send the call to the January 1st Call Handler; then, the January 1st CallHandler will do the same, since July 4th is marked as a holiday, it will forward the call to the Holiday Greeting which will send it to another CallHandler (July 4th). When the call gets to July 4th CallHandler it checks its schedule, since no holidays are marked on this particular CallHandler for July 4th, it will check whether it should play the Standard or Closed greeting, since this schedule was setup as 24/7, it will play the Standard Greeting of the July 4th CallHandler and stop hunting.
I hope you this helps you and feel free to shoot me with any questions you might have.
HTH
--espereir

Unity Connection 8.5 | Prepend Digits to Dialed Extensions overlap with an option in a call handler

Hi,
One customer has requested to set up a call handler in Unity Connection 8.5. The call handler should be capable of transfering calls to IP phones by dialing the last 4 digits of the IP phone's extension. The last 4 digits of the extension from all IP phones begin with 12. However, the customer is also requesting to have an option on the same call handler, so that if users would dial 1, the call is transfered to an IP phone
As you can see, there is an overlap between the last 4 digits of all IP phone's extensions, and an option in the call handler. Therefore, if I configure both options in the same call handler, when the users would press 1, Unity Connection will immediately send the call to the extension setup for option 1, even if the user wants to dial the last 4 digits of an IP phone extension
One possible solution would be to change the option in the call handler to transfer the call to an IP phone, to something different than 1. However, the customer wants to have this transfer to be performed when he press option 1
Is there any other possible solution for achieving this?
Thanks in advance

I don't see any problem, just don't check the option to ignore additional input when configuring caller input on digit 1
By default CUC will give you 1500 ms to dial for any other digits before trying to route the call with whatever you dialed.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

Unable to join Meet me conference from outside (DID) number

Hi,
Would anyone help to fix below!
What I'm trying to do?
- I'm working on meet me. Meet me pattern is 2222 and is obsolutly working fine internally. I used 2222 (DID) number so anyone can join conference from outside by dialing DID number 2222.
Issue.
- When i try to join Meet me conference by dialing DID it gives busy tone. Unable to join.
Is there something to do with transcode??
- I have already a transcode in CUCM and registered with Gateway
BR
Sam

Hi Manish,
Below debug details:
Calling number : 583125185 my Mobile
Called Number : 2287797 ( Meet me Pattern )
*Mar 26 08:46:08.176: //-1/E89798C3B280/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=583125185
----- ccCallInfo IE subfields -----
cisco-ani=583125185
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=2287797
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
*Mar 26 08:46:08.176: //-1/E89798C3B280/CCAPI/cc_api_call_setup_ind_common:
Interface=0x3C90FA24, Call Info(
Calling Number=583125185,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=2287797(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=5001, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=27631
*Mar 26 08:46:08.176: //-1/E89798C3B280/CCAPI/ccCheckClipClir:
In: Calling Number=583125185(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Mar 26 08:46:08.176: //-1/E89798C3B280/CCAPI/ccCheckClipClir:
Out: Calling Number=583125185(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Mar 26 08:46:08.176: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 26 08:46:08.176: :cc_get_feature_vsa malloc success
*Mar 26 08:46:08.176: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 26 08:46:08.176: cc_get_feature_vsa count is 3
*Mar 26 08:46:08.176: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 26 08:46:08.176: :FEATURE_VSA attributes are: feature_name:0,feature_time:1101743440,feature_id:1929
*Mar 26 08:46:08.176: //27631/E89798C3B280/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=583125185(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=2287797(TON=Unknown, NPI=Unknown))
*Mar 26 08:46:08.176: //27631/E89798C3B280/CCAPI/cc_process_call_setup_ind:
Event=0x3CB041D0
*Mar 26 08:46:08.176: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 2287797
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/ccCallSetContext:
Context=0x22CD2400
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 27631 with tag 5001 to app "_ManagedAppProcess_Default"
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=100, Params=0x22CDF0B0, Progress Indication=NULL(0)
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/ccCheckClipClir:
In: Calling Number=583125185(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/ccCheckClipClir:
Out: Calling Number=583125185(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/ccCallSetupRequest:
Destination Pattern=22877.., Called Number=2287797, Digit Strip=FALSE
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/ccCallSetupRequest:
Calling Number=583125185(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=2287797(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=
Account Number=583125185, Final Destination Flag=TRUE,
Guid=E89798C3-B3F9-11E3-B280-F273590C9FA3, Outgoing Dial-peer=100
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=583125185
----- ccCallInfo IE subfields -----
cisco-ani=583125185
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=2287797
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
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x3C90FA24, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=583125185,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=2287797(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=100, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
*Mar 26 08:46:08.180: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 26 08:46:08.180: :cc_get_feature_vsa malloc success
*Mar 26 08:46:08.180: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 26 08:46:08.180: cc_get_feature_vsa count is 4
*Mar 26 08:46:08.180: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 26 08:46:08.180: :FEATURE_VSA attributes are: feature_name:0,feature_time:1101742096,feature_id:1930
*Mar 26 08:46:08.180: //27632/E89798C3B280/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
*Mar 26 08:46:08.180: //27632/E89798C3B280/CCAPI/ccCallSetContext:
Context=0x22CDF060
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=100
*Mar 26 08:46:08.180: //27632/E89798C3B280/CCAPI/ccGetMediaClassTag:
media class tag 0
*Mar 26 08:46:08.180: //27632/E89798C3B280/CCAPI/ccSetMediaclassIp2ipTags:
media class tags set: NR 0, ASP 0
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/ccGetMediaClassTag:
media class tag 0
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/ccSetMediaclassIp2ipTags:
media class tags set: NR 0, ASP 0
*Mar 26 08:46:08.180: //27632/E89798C3B280/CCAPI/ccGet_xc_nr_asp_info:
media class tags: NR 0, ASP 0
*Mar 26 08:46:08.180: //27631/E89798C3B280/CCAPI/ccGet_xc_nr_asp_info:
media class tags: NR 0, ASP 0
*Mar 26 08:46:08.180: //27632/E89798C3B280/CCAPI/cc_api_event_indication:
Event=188, Call Id=27632
*Mar 26 08:46:08.180: //27632/E89798C3B280/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Mar 26 08:46:08.180: //27631/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.180: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.180: //27632/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.180: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.184: //27631/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.184: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.184: //27631/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.184: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.184: //27632/E89798C3B280/CCAPI/cc_api_call_proceeding:
Interface=0x3C90FA24, Progress Indication=NULL(0)
*Mar 26 08:46:08.184: //27632/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.184: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.184: //27631/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.184: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.184: //27632/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.184: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.184: //27632/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.184: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.200: //27632/E89798C3B280/CCAPI/cc_api_call_disconnected:
Cause Value=1, Interface=0x3C90FA24, Call Id=27632
*Mar 26 08:46:08.200: //27632/E89798C3B280/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=1, Retry Count=0)
*Mar 26 08:46:08.200: //27631/E89798C3B280/CCAPI/ccCallReleaseResources:
release reserved xcoding resource.
*Mar 26 08:46:08.200: //27632/E89798C3B280/CCAPI/ccCallSetAAA_Accounting:
Accounting=1, Call Id=27632
*Mar 26 08:46:08.200: //27632/E89798C3B280/CCAPI/ccCallDisconnect:
Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=1)
*Mar 26 08:46:08.200: //27632/E89798C3B280/CCAPI/ccCallDisconnect:
Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
*Mar 26 08:46:08.200: //27632/E89798C3B280/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x3C90FA24, Tag=0x0, Call Id=27632,
Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)
*Mar 26 08:46:08.200: //27632/E89798C3B280/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
*Mar 26 08:46:08.200: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 26 08:46:08.200: :cc_free_feature_vsa freeing 41AB4008
*Mar 26 08:46:08.200: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 26 08:46:08.200: vsacount in free is 3
*Mar 26 08:46:08.200: //27631/E89798C3B280/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=5000, Params=0x22CCBE20, Progress Indication=NULL(0)
*Mar 26 08:46:08.200: //27631/E89798C3B280/CCAPI/ccCheckClipClir:
In: Calling Number=583125185(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Mar 26 08:46:08.204: //27631/E89798C3B280/CCAPI/ccCheckClipClir:
Out: Calling Number=583125185(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Mar 26 08:46:08.204: //27631/E89798C3B280/CCAPI/ccCallSetupRequest:
Destination Pattern=.T, Called Number=2287797, Digit Strip=FALSE
*Mar 26 08:46:08.204: //27631/E89798C3B280/CCAPI/ccCallSetupRequest:
Calling Number=583125185(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=2287797(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=
Account Number=583125185, Final Destination Flag=TRUE,
Guid=E89798C3-B3F9-11E3-B280-F273590C9FA3, Outgoing Dial-peer=5000
*Mar 26 08:46:08.204: //27631/E89798C3B280/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=583125185
----- ccCallInfo IE subfields -----
cisco-ani=583125185
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=2287797
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
*Mar 26 08:46:08.204: //27631/E89798C3B280/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x3C90FA24, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=583125185,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=2287797(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=5000, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
*Mar 26 08:46:08.204: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 26 08:46:08.204: :cc_get_feature_vsa malloc success
*Mar 26 08:46:08.204: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 26 08:46:08.204: cc_get_feature_vsa count is 4
*Mar 26 08:46:08.204: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 26 08:46:08.204: :FEATURE_VSA attributes are: feature_name:0,feature_time:1101742096,feature_id:1931
*Mar 26 08:46:08.204: //27633/E89798C3B280/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
*Mar 26 08:46:08.204: //27633/E89798C3B280/CCAPI/ccCallSetContext:
Context=0x22CCBDD0
*Mar 26 08:46:08.204: //27631/E89798C3B280/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=5000
*Mar 26 08:46:08.204: //27633/E89798C3B280/CCAPI/ccGetMediaClassTag:
media class tag 0
*Mar 26 08:46:08.204: //27633/E89798C3B280/CCAPI/ccSetMediaclassIp2ipTags:
media class tags set: NR 0, ASP 0
*Mar 26 08:46:08.204: //27631/E89798C3B280/CCAPI/ccGetMediaClassTag:
media class tag 0
*Mar 26 08:46:08.204: //27631/E89798C3B280/CCAPI/ccSetMediaclassIp2ipTags:
media class tags set: NR 0, ASP 0
*Mar 26 08:46:08.204: //27633/E89798C3B280/CCAPI/ccGet_xc_nr_asp_info:
media class tags: NR 0, ASP 0
*Mar 26 08:46:08.204: //27631/E89798C3B280/CCAPI/ccGet_xc_nr_asp_info:
media class tags: NR 0, ASP 0
*Mar 26 08:46:08.204: //27633/E89798C3B280/CCAPI/cc_api_event_indication:
Event=188, Call Id=27633
*Mar 26 08:46:08.204: //27633/E89798C3B280/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Mar 26 08:46:08.204: //27631/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.204: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.204: //27633/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.204: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.204: //27631/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.204: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.204: //27631/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.204: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.204: //27633/E89798C3B280/CCAPI/cc_api_call_proceeding:
Interface=0x3C90FA24, Progress Indication=NULL(0)
*Mar 26 08:46:08.208: //27633/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.208: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.208: //27631/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.208: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.208: //27633/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.208: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.208: //27633/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Mar 26 08:46:08.208: cc_api_get_xcode_stream : 4819
*Mar 26 08:46:08.280: //27633/E89798C3B280/CCAPI/cc_api_call_disconnected:
Cause Value=28, Interface=0x3C90FA24, Call Id=27633
*Mar 26 08:46:08.280: //27633/E89798C3B280/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=28, Retry Count=0)
*Mar 26 08:46:08.280: //27631/E89798C3B280/CCAPI/ccCallReleaseResources:
release reserved xcoding resource.
*Mar 26 08:46:08.280: //27633/E89798C3B280/CCAPI/ccCallSetAAA_Accounting:
Accounting=0, Call Id=27633
*Mar 26 08:46:08.280: //27633/E89798C3B280/CCAPI/ccCallDisconnect:
Cause Value=28, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=28)
*Mar 26 08:46:08.280: //27633/E89798C3B280/CCAPI/ccCallDisconnect:
Cause Value=28, Call Entry(Responsed=TRUE, Cause Value=28)
*Mar 26 08:46:08.284: //27633/E89798C3B280/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x3C90FA24, Tag=0x0, Call Id=27633,
Call Entry(Disconnect Cause=28, Voice Class Cause Code=0, Retry Count=0)
*Mar 26 08:46:08.284: //27633/E89798C3B280/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
*Mar 26 08:46:08.284: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 26 08:46:08.284: :cc_free_feature_vsa freeing 41AB4008
*Mar 26 08:46:08.284: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 26 08:46:08.284: vsacount in free is 3
*Mar 26 08:46:08.284: //27631/E89798C3B280/CCAPI/ccCallDisconnect:
Cause Value=28, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Mar 26 08:46:08.284: //27631/E89798C3B280/CCAPI/ccCallDisconnect:
Cause Value=28, Call Entry(Responsed=TRUE, Cause Value=28)
*Mar 26 08:46:08.300: //27631/E89798C3B280/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x3C90FA24, Tag=0x0, Call Id=27631,
Call Entry(Disconnect Cause=28, Voice Class Cause Code=0, Retry Count=0)
*Mar 26 08:46:08.300: //27631/E89798C3B280/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
*Mar 26 08:46:08.300: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 26 08:46:08.300: :cc_free_feature_vsa freeing 41AB4548
*Mar 26 08:46:08.300: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
BR
Sami

Unity Connection 9.1 - System Call Handler Question

Hi,
I would like to use Unity Conncetion 9.1 to transfer calls to different extensions depending on the time of day:
For example; between 07.00 and 19.00, do not play a greeting and transfer to an extension (no mailbox), outside of the times above, again do not play a greeting and transfer to a different extension.
I have configured a CTI RP and set the extension to forward all to VM. The system call handler has an active schedule of 07.00-19.00 and has a standard transfer rule set up to transfer the calls to the relevant extension. I have also configured a closed transfer rule to transfer the call to the alternative extension after 19.00. The issue I have is that when dial the original extension I get the standard greeting. How do i get Unity to transfer the call without playing the greeting?
Thanks,
Phil

Hi Phil,
You need to:
1) enable the closed greeting
2) set the standard and closed greetings to "callers hear nothing"
3) in the standard and closed greetings, after the greeting, set it to call handler, select the same call handler, and set it to attempt transfer.
4) make sure the standard and closed transfer rules are enabled, set the respective extensions to transfer to, set them to release to switch, and uncheck "Play the 'Wait While I Transfer Your Call' Prompt" for both of them.

Allowing caller to route to an extension while listening to call handler greeting

We have a Unity 7.x server. We want to configure it such that when an outside caller dials into the main line (hits a call handler), they can tap in an extension at any time during the playback of the opening greeting and be transferred to that extension. Not just the voice mailbox for that extension. They want the call to be transferred to that extension. Right now it is not working. Instead the call just goes right to the voice mailbox of the extension that is entered during playback of the opening greeting. In the "During greeting:" section of the call handler, the "Allow caller input" box is checked. Under the "Caller Input", the "Allow callers to dial an extension during greeting" is also checked. What's missing?

Go to the subscribers and under transfer settings set it to yes, ring subscriber phone. The default behavior is to send calls to VM and not ring the phone.
For this I assume you mean they already have a subscriber, for non-subscriber DNs you need a call system transfer.
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk

Call forward to unity connection call handler

have the following setup:
cucm 8.6
cuc 8.6
a cti route point (DN 1000 )with forward all to unity connection call handler.
phone users who choose to have some quiet time can forward all their calls to the cti rp.
1. i need to allow only specfic DNs the ability to forward to the cti rp.
2. at times that the allowed DNs didnt choose to forward to cti rp then normal behaviour should occur - busy or unanswerd calls should go to DNs personal voice mail.
thanks

You can assign a specific partition to cti-rp and assign it to a CSS and this CSS should be assigned to those users (at call forward settings in line) only to whom you are trying to give this facility and for other users you can
or you can create a TP for it with specific partition and set the called number as per voice mail.
Suresh

Call Handler Transfer to Meet Me Conference

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