Conference Call Single Disconnect

How can you be on a 3 way (or more) conference call, dial a number & hang up the number you dialed while staying on the original conference call?

it has nothing to do with your phone model. this is just part of verizon. you can't even do this on the 4g lte network with a brand new iphone 5 or samsung galaxy 3s.
i have to do this every day...here's a daily life example from my sales job: I'm team-calling a sales prospect with a colleague. I initiate a 3-way call to a prospect, get their voicemail, then have to hang up on my colleague to end the call to the prospect.

Similar Messages

Problem disconnecting a participant in call waiting and conference calls.

I have a Z10 running 10.3.2 on Verizon. I have not found a way to hang up on one caller in a call waiting or conference call without disconnecting all calls. The only option the screen offers is the end call button. I can switch between callers and merge calls just fine but if I try to end one of those calls, it will disconnect all participants and end the call completely. Does anybody have a solution to this?

Ok. So I completely forgot about this until the flag I put on the problem came up.
So it is working now and as you suggested I called Telusand they "pushed" the SIM and refreshed the switch "in the back". Have no clue what either of those actions entails the critical item being that it now works.
Thanks

Conference call and get disconnect tone

Hi,
I had a issues as below:-
Now have 5 party in conference call, 2 from internal and 3 call to PSTN's user. Suddenly one of the PSTN user call are drop then the remaining user in conference call are hearding the disconnect tone for few minutes. May I know how to shorted the disconnect tone or totally no disconnect tone after the PSTN user disconnect? Please let me know how to solved this.
I using CUCM 6.1 and conference station 7936.
Thank you.

Hey! Most likely it is an engineering issue, should be fixed soon with the newest update.
Check this related thread: How to add caller to Facetime audio call in ios8?

How can I disconnect one caller from a conference call?

I am not able to disconnect one call from a conference call. I have tried everything and do not wish to end both calls, just one.

I'd love to clarify this information bdgooden. Unfortunately, your only option is to disconnect all lines. The other parties on the phone call can disconnect the line to end the call, but you only have the option of disconnecting all calls or waiting for the other party to end.
Thank you,
MichelleH_VZW
Follow us on Twitter @VZWSupport

Conference Call Disconnects At Same Point Every Time

Each time I make a conference call on my iPhone, it disconnects after no more than 15 minutes. It just hangs up. I'm not sure what happens. I don't have any other problems with it. Is anyone else having this problem when making a conference call?

My husband has been experiencing the same problem. I wonder if there is a standard time-out because it always seems to be after 15 minutes.

Starkey Halo - Disconnecting from Conference Calls?

I use my Starkey Halo i110's as my telephone headset. (It's the only way I can understand others on the phone.)
I work from home. I use our company's conference calling provider extensively; sometimes 5-6 times a day or more. Occasionally, I will get dropped from a conference call. The conference calling host sees a normal call disconnect from Verizon Wireless (my wireless provider). Their carrier (AT&T) also sees a normal call disconnect from Verizon. Verizon sees a disconnect, but upstream from them. Verizon also claims that I am at the edge of service which I would dispute. There aren't many places in Central Florida where you can avoid seeing a cell tower. I am just SE of Tampa, about 20 miles as the crow flies from the airport.
I have tested with both an iPhone 5C and 5S, both with Verizon Wireless. I have had conference calls get dropped on both phones. In 3+ years, I have had no other dropped cell calls. Only the conference calls get dropped. I have had the dropped calls with the last version (or 2?) of iOS 7 and with iOS 8.0 and iOS 8.0.2.
The problem started about when I started wearing the Halos. That's been nearly four months now and it just occurred to me today that the Halos could be a significant part of the problem (I never claimed to be smart or quick....).
Do any other iPhone & Starkey users experience any dropped calls?

I received a reply fro Starkey tech support that told me the Halos couldn't issue an "end call" command. Consequently, the problem is not with the Halos.
I am relieved.

External callers unable to join conference calls

Scenario: A conference call is set up using the scheduler. Lync users are able to join the conference calls. Users from outside PSTN are unable to join the call. They get to the attendant, and are told they are being joined to the
conference, then get a message "Sorry, I can't seem to connect you to your meeting right now. Goodbye." and the call disconnects.
Looking at call traces, I see the following error messages in the CAAServer log:
Component: CAAServer
Level: TL_ERROR
Flag: TF_DIAG
Function: CaaCall.EndTransfer
Source: caacall.cs(3631)
Local Time: 08/26/2014-18:28:36.161
Sequence# : 0001BE31
CorrelationId : 64624070
ThreadId : 0BA8
ProcessId : 0BB0
CpuId : 0
Original Log Entry :
TL_ERROR(TF_DIAG) [0]0BB0.0BA8::08/26/2014-22:28:36.161.0001be31 (CAAServer,CaaCall.EndTransfer:caacall.cs(3631))[64624070]ConferenceFailureException. ConferenceID=[91127] InstanceID=[b64b9467-db72-41bb-a2b5-673d76777ea4]
Component: CAAServer
Level: TL_ERROR
Flag: TF_DIAG
Function: CaaCall.HandleUnexpectedException
Source: caacallerrorhandling.cs(51)
Local Time: 08/26/2014-18:28:36.226
Sequence# : 0001D265
CorrelationId : 64624070
ThreadId : 0BA8
ProcessId : 0BB0
CpuId : 2
Original Log Entry :
TL_ERROR(TF_DIAG) [2]0BB0.0BA8::08/26/2014-22:28:36.226.0001d265 (CAAServer,CaaCall.HandleUnexpectedException:caacallerrorhandling.cs(51))[64624070][Enter] exception=[Microsoft.Rtc.Collaboration.ConferenceFailureException:The operation failed due to a response
from the server. For more information, examine the properties on the exception and inner exception.
   at Microsoft.Rtc.Signaling.SipAsyncResult`1.ThrowIfFailed()
   at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult result)
   at Microsoft.Rtc.Collaboration.McuSession.EndSendCommandInternal(IAsyncResult result)
   at Microsoft.Rtc.Collaboration.AudioVideo.AudioVideoMcuSession.EndTransfer(IAsyncResult result)
   at Microsoft.LiveServer.Caa.CaaCall.EndTransfer(IAsyncResult asyncResult, Boolean& retry, Exception& caught)
Detected at System.Environment.get_StackTrace()
   at Microsoft.Rtc.Collaboration.ConferenceFailureException..ctor(String message, Exception innerException)
   at Microsoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.CreateConferenceFailureException(ConferenceCommandResponse commandResponse, Exception innerException)
   at Microsoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.ProcessCccpResponse(SipMessageData messageData, responsetype response, Boolean& isPendingResponse)
   at Microsoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.ProcessStatusMessage(SipMessageData statusMessageData, responsetype response)
   at Microsoft.Rtc.Collaboration.Conferencing.StatusMessageReceivedWorkItem.Process()
   at Microsoft.Rtc.Signaling.AsyncWorkitemQueue.ProcessItems()
   at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessing()
   at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessingCallback(Object state)
   at Microsoft.Rtc.Signaling.QueueWorkItemState.ExecuteWrappedMethod(WaitCallback method, Object state)
   at System.Threading.ExecutionContext.Run(ExecutionContext executionContext, ContextCallback callback, Object state)
   at System.Threading._ThreadPoolWaitCallback.PerformWaitCallbackInternal(_ThreadPoolWaitCallback tpWaitCallBack)
   at System.Threading._ThreadPoolWaitCallback.PerformWaitCallback(Object state)] m_conferenceID=[91127] m_conferenceUri=[sip:[email protected];gruu;opaque=app:conf:focus:id:4TPY99KF] InstanceID=[b64b9467-db72-41bb-a2b5-673d76777ea4]
Component: CAAServer
Level: TL_ERROR
Flag: TF_DIAG
Function: CaaCall.HandleUnexpectedException
Source: caacallerrorhandling.cs(51)
Local Time: 08/26/2014-18:28:36.226
Sequence# : 0001D265
CorrelationId : 64624070
ThreadId : 0BA8
ProcessId : 0BB0
CpuId : 2
Original Log Entry :
TL_ERROR(TF_DIAG) [2]0BB0.0BA8::08/26/2014-22:28:36.226.0001d265 (CAAServer,CaaCall.HandleUnexpectedException:caacallerrorhandling.cs(51))[64624070][Enter] exception=[Microsoft.Rtc.Collaboration.ConferenceFailureException:The operation failed due to a response
from the server. For more information, examine the properties on the exception and inner exception.
   at Microsoft.Rtc.Signaling.SipAsyncResult`1.ThrowIfFailed()
   at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult result)
   at Microsoft.Rtc.Collaboration.McuSession.EndSendCommandInternal(IAsyncResult result)
   at Microsoft.Rtc.Collaboration.AudioVideo.AudioVideoMcuSession.EndTransfer(IAsyncResult result)
   at Microsoft.LiveServer.Caa.CaaCall.EndTransfer(IAsyncResult asyncResult, Boolean& retry, Exception& caught)
Detected at System.Environment.get_StackTrace()
   at Microsoft.Rtc.Collaboration.ConferenceFailureException..ctor(String message, Exception innerException)
   at Microsoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.CreateConferenceFailureException(ConferenceCommandResponse commandResponse, Exception innerException)
   at Microsoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.ProcessCccpResponse(SipMessageData messageData, responsetype response, Boolean& isPendingResponse)
   at Microsoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.ProcessStatusMessage(SipMessageData statusMessageData, responsetype response)
   at Microsoft.Rtc.Collaboration.Conferencing.StatusMessageReceivedWorkItem.Process()
   at Microsoft.Rtc.Signaling.AsyncWorkitemQueue.ProcessItems()
   at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessing()
   at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessingCallback(Object state)
   at Microsoft.Rtc.Signaling.QueueWorkItemState.ExecuteWrappedMethod(WaitCallback method, Object state)
   at System.Threading.ExecutionContext.Run(ExecutionContext executionContext, ContextCallback callback, Object state)
   at System.Threading._ThreadPoolWaitCallback.PerformWaitCallbackInternal(_ThreadPoolWaitCallback tpWaitCallBack)
   at System.Threading._ThreadPoolWaitCallback.PerformWaitCallback(Object state)] m_conferenceID=[91127] m_conferenceUri=[sip:[email protected];gruu;opaque=app:conf:focus:id:4TPY99KF] InstanceID=[b64b9467-db72-41bb-a2b5-673d76777ea4]
Component: CAAServer
Level: TL_ERROR
Flag: TF_DIAG
Function: CaaCall.TransferCompletedQueued
Source: caacall.cs(3739)
Local Time: 08/26/2014-18:28:36.231
Sequence# : 0001D269
CorrelationId : 64624070
ThreadId : 0BA8
ProcessId : 0BB0
CpuId : 2
Original Log Entry :
TL_ERROR(TF_DIAG) [2]0BB0.0BA8::08/26/2014-22:28:36.231.0001d269 (CAAServer,CaaCall.TransferCompletedQueued:caacall.cs(3739))[64624070]Call Transfer Failed. InstanceID=[b64b9467-db72-41bb-a2b5-673d76777ea4]
The unexpected error entry includes this information in the diagnosticUtils section:
Component: CAAServer
Level: TL_INFO
Flag: TF_DIAG
Function: DiagnosticUtils.TryCreateSignalingHeader
Source: diagnosticutils.cs(269)
Local Time: 08/26/2014-18:28:36.231
Sequence# : 0001D268
CorrelationId :
ThreadId : 0BA8
ProcessId : 0BB0
CpuId : 2
Original Log Entry :
TL_INFO(TF_DIAG) [2]0BB0.0BA8::08/26/2014-22:28:36.231.0001d268 (CAAServer,DiagnosticUtils.TryCreateSignalingHeader:diagnosticutils.cs(269))Creating SignalingHeader for InstanceID=[b64b9467-db72-41bb-a2b5-673d76777ea4], header-name=[ms-diagnostics], header-value=[10010;Reason="Gateway
side Media negotiation failed";Source="LYNC-FE.itprocare.com";component="MediationServer";sipresponsetext="Invite with Replaces failed because Gateway side reinvite failed.";DialogID="f94f35b9-7f5d-4bcd-94a8-de0fb2d8c4c6;df1bec7e24;24010080"]
Which matches up with an error in the SIP trace:
TL_INFO(TF_PROTOCOL) [0]0DEC.0A48::08/26/2014-22:28:36.151.0001ac66 (SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record
Trace-Correlation-Id: 3329945913
Instance-Id: 00000688
Direction: outgoing
Peer: lync-fe.itprocare.com:51307
Message-Type: response
Start-Line: SIP/2.0 491 Invite with Replaces failed because Gateway side reinvite failed.
From: <sip:[email protected];gruu;opaque=app:conf:audio-video:id:4TPY99KF>;tag=26f350618b;epid=6DF0663499
To: <sip:[email protected];gruu;opaque=srvr:MediationServer:BMFt1DuKo1KYsGtnqeobCwAA;grid=9d179e4c3aeb4fd7aedd36d7853ad98b>;epid=26F55811B8;tag=2e6beef249
CSeq: 5 INVITE
Call-ID: af099053-d8aa-4ca4-9820-936e8522611c
Via: SIP/2.0/TLS 10.160.1.47:51307;branch=z9hG4bKeb2e7ed6;ms-received-port=51307;ms-received-cid=9B00
CONTENT-LENGTH: 0
P-ASSERTED-IDENTITY: <sip:5853307343;[email protected];user=phone>
SERVER: RTCC/4.0.0.0 MediationServer
ms-diagnostics: 10010;source="LYNC-FE.itprocare.com";reason="Gateway side Media negotiation failed";component="MediationServer";SipResponseText="Invite with Replaces failed because Gateway side reinvite failed."
ms-diagnostics-public: 10010;reason="Gateway side Media negotiation failed";component="MediationServer";SipResponseText="Invite with Replaces failed because Gateway side reinvite failed."
ms-endpoint-location-data: NetworkScope;ms-media-location-type=intranet
Message-Body: –
$$end_record
The trunks are configured with REFER off and MediaBypass Off.
We have recently moved to a direct SIP trunk from a vendor on Microsoft's Certified list.
Lync version is 2010, with the latest CUs applied. All other calling appears to be working correctly. The certificate on the servers are using the SHA1 algorithm (I have seen some similar issues discussed if this was not the case.)
At this point, I have reached the end of my immediate troubleshooting skills with this system. Can anyone offer any suggestions as to what might be going on here?
Thanks for any help.
-Tim

Hi,
Please check if the default gateway associated to the Mediation Server is up or not.
Please check if Media traffic on the Gateway be blocked with the issue of wrong encryption. Make sure it is using SRTP (not RTP). If you use RTP, please change it and then test again.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

How to hang up on conference call on CDMA without hanging up on the first one?

Is it possible to start a conference call and then disconnect with the second person called and stay on with the first on CDMA network?

When someone hangs up it still says conference call and then I am no longer able to make a new 3 way call.... the other person could be there on the phone still but it makes me wonder if i am being charged mins for the other person still cause it still shows conference still. Kinda bugs me. i came from T MOBILE and I could join calls and manage calls etc but on Verizon i cant do this....

Do I get full conference call management options on iPhone 6 model A1549 if using it with GSM carriers?

I have iPhone 6 model A1549, bought from and currently using with Verizon. You all must be aware that we can use the same phone for GSM carriers like at&t as well. Now we all know that some of the features related to conference calling are not available on CDMA carriers, like we cant have more than 2 callers, and we can't manage conference calls by making the conversation with one party private for some time and again switching back to conference mode, disconnecting a particular party instead of having to end the entire call etc. reference link- iPhone: Understanding phone features - Apple Support These features are available on GSM models and carriers, I just wanted to ask if I use a GSM SIM from a GSM carrier on this model, do I get these additional conference call management options or I am restricted to Verizon like experience while being on at&t (and while using the at&t SIM in this model)

Yes, I am dealing with this right now on at&t with a 6 plus. I've been down the same road as you. I got it to work with a windows phone yesterday but not my 6 plus when I switched back to the iPhone. My wifes 6 works fine. I believe it is an at&t issue and not an 8.3 issue. After dealing through several levels of support at at&t, they are telling me it has to do with their transition to hd voice. Some people are having this issue because they are updating their system and some peoples phone profile, on phones that have hd voice capabilities, are not getting added to their line and call forwarding will not work without it. I have a trouble ticket open to their engineering dept and they promise it will work for me on or before the 24th of april. Engineering has to build my phones profile so call forwarding will work with my iPhone.
This is frustrating because like you, I need to forward my calls because I have no signal in my office and need to forward to another phone so I can get my calls. If at&t would quit dragging their feet and enable wifi calling, this wouldn't be an issue for me. I would suggest you put more pressure on at&t and escalate this until it gets resolved. Or put your sim in a phone that does not have hd voice capability and see if it works then.
Good luck. I'm keeping my fingers crossed this will be fixed for me on or before the 24th. We'll see.
EDIT: I see you did put it in another phone and it worked. That backs up that the issue is at&t and their hd voice upgrades. They had no explanation as to why some work and some don't, but have them make sure a profile for your phone is on the line. If not, they'll need to build one for the phone so that it''l work. At least that's the story for now. Like I said, I'll know more in a day or two.

How do I hang up on a conference call

Need Help Here! How do I hang up on a conference 3-way call without completly diconnct the call?

Tap on the 'i' next to the word Conference at the top of the screen, and choose which call to end.
If this option doesn't appear, your carrier may not support disconnecting from individual calls in a conference call, and you may have to ask the relevant party to disconnect.

Conference call dropped

We were able to use conference call for 20 senior management in the monthly meeting. In the last meeting, 8 conference calls dropped middle of meeting.
Nothing changes in existing configuration. I have checked in CUCM Server, System > Service Parameter > Call Manager > Clusterwide parameter>
Drop Ad Hoc Conferenece : Never
Advanced Ad Hoc Conference Enable : True
Great appreciated someone advice me above issue.

Does the call actually end when the conference drops or does the audio just stop (one way or no way audio)? If you have Call Detailed Records (CDR) enabled you can check the disconnect reason of the call. If you don't and you have detailed call manager traces you can take a look there to see why the calls dropped if they actually cleared the call.

3-Way Calling/Conference Call needs to be imrpoved asap!!

Well come on verizon please fix this problem with the 3-way calling. Im only able to do one 3-way and thats all. only one time during the call. if the other line ends i cannot make another 3-way. I dont even have the option to end one line of the conference call. Im just stuck in conference mode. You should be able to hit conference and then something will show up and let you end one line. I used an AT&T iphone 5 and it works so much better. You have the option to end whichever line you want and you can add another line if u wanted to. Verizon you should be able to make it work like the At&t iphone. With the old basic phone im able to make as many conference calls as I like by just keep hitting the send button. I didnt pay $350 for an iphone 5 that doesnt have good 3-way calling features. Please improve ASAP. It should be like the AT&T iphone 5. When is the fix gonna happen becuase it has been like this for a long time.

Huh! You're right (even though that's not claimed in the specs).
I still detest the idea of having to run twin modems/radios to to voice & data.
There are 3 things that really bug me & may lead to me leaving Verizon:
1. The stated issue with multi-party calls (which this thread was about before Ohio & I hijacked it for a iPhone/Droid voice & data discussion.)
Even though I don't often have 6 lines going at once, the ability to merge incoming calls seems like a really basic functionality that I used to use quite frequently.
2. When I dial conference lines # have the passcode dialed as part of the number I used to be able to put: phonenumber,passcode.
This on AT&T (or any GSM or UMTS network) dials the number (silently), waits until it connects, dials the passcode.
On verizon, this seems to dial the number silently, start dialing the passcode to a ringing line.
OK: add more pauses...
but now it seems to dial the passcode silently (like with the phone number, which the system accepts) AND with tone. This would be OK if they were in sync, but they're not. I get: Thanks for calling, please enter your passcode: "beep!" Thanks, you're now connected. --and the rest of the people on my conference call hear the remaining beeps and boops until the passcode is complete.
#VZFail
3. Voice & data.
It's actually the one I care the least about, but it'd be nice to have a call AND use my phone for a mobile hot-spot. I know VoLTE will fix this, but I'm shocked to hear AT&T may beat VZ to the punch on this since Verizon has had LTE since just about forever.
(In fairness, AT&T has been less than ambitious to deliver LTE because their "3G" is 72Mbit capable & LTE is 100Mbit. VZ's 3G is single digit Mbit & LTE represents a huge jump. --and AT&T's 3G network can do voice + data with 1 radio, hold 6 calls at once, merge/unmerge any line at will. --you just get dropped calls.)
*sigh* Why can't I just have it all?!

Conference calls technical issues

Hi. I wonder if you can help me. My question and issue is perhaps better dealt with via phone, however, I can give it a shot here: I make group calls from my second account. We have a process whereby the host for the call rotates every week (on average).
Last couple of times someone else made those conference calls, we had a lot of technical difficulties, dropped calls, muted sharing, disconnections and such.
Why is this happening? How can we overcome our challenge?
Thank you in advance for your support.

If asking for help you need to provide full details and not just the problem.
Call control agent?
version?
If CUCM, SW CFB/HW CFB?
By what you mention it sound like codec issues, so start looking into that in first place.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

Number of concurrent a/v conference calls, and its bandwidth consumed.

I have two central sites and multiple branch sites. Each central site has one Front End pool. we are planning to replace Lync 2013 servers for hosting meetings, conferences ( audio, video and web)
My manager wants to know the number of concurrent audio/Video conference calls that our Lync Front End pools can handle and the bandwidth consumed by these calls. Can anyone here help with this or point me to where to find the right tool. I have tried using
Bandwidth calculator, it was not working well for me.

For Audio, assuming G722 (without FEC) it will be about 96Kbps per stream. So in theory that would be
over 1000 calls on 100Mb, however that doesn't take into account any other traffic that would be on the link.
For H.264 Video it can be anywhere up to 4000Kbps (HD), but it's unlikely that all your end points would be running 1920x1080 resolution and they will all negotiate differently. So without knowing a lot about your environment I would say on average (which
is just that between min to max around 50 video streams) or 128-250 streams at lower resolutions - so traffic through phones and slow connections (320x180-16:9/ 212x160-4:3 to 640x360-16:9/640x480-4:3) and
25-50 streams at the higher resolutions. So HD cameras and rooms systems, etc. Once again not taking into account other things on the network.
With regards to the pool themselves, follow the hardware specifications required which you can check using the capacity planning tool http://www.microsoft.com/en-us/download/details.aspx?id=36828 the
supported maximum however for a large conference on a single pool is 250 participants.
As I mentioned it's almost impossible to give definitive numbers because there are so many variables, but hopefully this helps a little.
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Call gets disconnected after playing the menu from CVP

Hello,
When I make a call, the call is going to the ICM and then it is coming to the Call server then to the VXML gateway and then it goes to the VXML server and plays the promt.
The problem i am facing is that once the menu audio is played the call gets disconnected. It doesnt let me choose any options. I feel it is not able to identify the my inputs
I have some debug outputs
Mar 1 01:09:03.095: //-1/C113522D805A/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=1000
   ----- ccCallInfo IE subfields -----
   cisco-ani=sip:[email protected]:5060
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=sip:[email protected];transport=udp
   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 1 01:09:03.103: //-1/C113522D805A/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x67213130, Call Info(
   Calling Number=sip:[email protected]:5060,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=sip:[email protected];transport=udp(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Un
R1#known, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=811, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=65
*Mar 1 01:09:03.103: //-1/C113522D805A/CCAPI/ccCheckClipClir:
   In: Calling Number=sip:[email protected]:5060(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Mar 1 01:09:03.103: //-1/C113522D805A/CCAPI/ccCheckClipClir:
   Out: Calling Number=sip:[email protected]:5060(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Mar 1 01:09:03.103: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 1 01:09:03.103: :cc_get_feature_vsa malloc success
*Mar 1 01:09:03.103: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 1 01:09:03.103: cc_get_feature_vsa count is 1
*Mar 1 01:09:03.103: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 1 01:09:03.103: :FEATURE_VSA attributes are: feature_name:0,feature_time:1732532552,feature_id:23
*Mar 1 01:09:03.103: //65/C113522D805A/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=(TON=Unknown, NPI=Unknown))
*Mar 1 01:09:03.111: //65/C113522D805A/CCAPI/cc_process_call_setup_ind:
   Event=0x67452800
*Mar 1 01:09:03.119: //65/C113522D805A/CCAPI/ccCallSetContext:
   Context=0x65097114
*Mar 1 01:09:03.119: //65/C113522D805A/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 65 with tag 811 to app "_ManagedAppProcess_bootstrap"
*Mar 1 01:09:03.143: //65/C113522D805A/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
*Mar 1 01:09:03.147: //65/C113522D805A/CCAPI/ccCallConnect:
   Progress Indication=NULL(0), Data Bitmask=0x0
*Mar 1 01:09:03.151: //65/C113522D805A/CCAPI/ccCallConnect:
   Call Entry(Connected=TRUE, Responsed=TRUE)
*Mar 1 01:09:03.239: //65/C113522D805A/CCAPI/cc_api_event_indication:
   Event=141, Call Id=65
*Mar 1 01:09:03.243: //65/C113522D805A/CCAPI/cc_api_event_indication:
   Event Is Sent To Conferenced SPI(s) Directly
*Mar 1 01:09:03.243: //65/C113522D805A/CCAPI/cc_api_caps_ind:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=65,
   Caps(Codec=0x0, Fax Rate=0x2, Vad=0x2,
   Modem=0x0, Codec Bytes=20, Signal Type=2)
*Mar 1 01:09:03.243: //65/C113522D805A/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=250(ms), Fax Nom=300(ms))
*Mar 1 01:09:04.163: //65/C113522D805A/CCAPI/ccCallHandoff:
   Silent=FALSE, Application=0x6773AB4C, Conference Id=0xFFFFFFFF
*Mar 1 01:09:04.295: //65/C113522D805A/CCAPI/ccSetDigitTimeouts:
   Initial Digit Timeout=0(ms), Inter Digit Timeout=0(ms)
*Mar 1 01:09:04.295: //65/C113522D805A/CCAPI/ccSetDigitTimeouts:
   Call Entry(Inter Digit Timeout=0(ms), Initial Digit Timeout=0(ms))
*Mar 1 01:09:04.295: //65/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
   (callID=0x41, digit_event=0x1, enable=TRUE, consume=FALSE)
*Mar 1 01:09:04.299: //65/C113522D805A/CCAPI/ccCallReportDigits:
   Enabled=TRUE, Call Id=65
*Mar 1 01:09:04.303: //65/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
   (vdbPtr=0x67213130, callID=0x41, disp=-4, digit_event=0x1, enable=TRUE, consume=FALSE)
*Mar 1 01:09:04.307: //65/C113522D805A/CCAPI/cc_api_call_report_digits_done:
   Enabled=TRUE, Disposition=0xFFFFFFFC, Interface=0x67213130, Call Id=65
*Mar 1 01:09:04.311: //65/C113522D805A/CCAPI/cc_api_call_report_digits_done:
   Call Entry(Initial Digit Timeout=0(ms), Inter Digit Timeout=0(ms))
*Mar 1 01:09:05.291: //65/C113522D805A/CCAPI/ccAssociateStream:
   Coder=5, DTMF Relay=4, Vad=0,
   Record Function=0x0, Event Queue=0x66954714, Stream Context=678B0364,
   Record Context=0x0, Stream Call Id=66, Call Id=65
*Mar 1 01:09:05.295: //65/C113522D805A/CCAPI/ccAssociateStream:
   Call Entry(Stream Status=2, Digit Enable=TRUE)
*Mar 1 01:09:05.327: //65/C113522D805A/CCAPI/cc_api_call_associated:
   Interface=0x0, CallId=65, Disposition=0
   Play Function=0x62C2DFA8, Codec=0x5, Vad=0x0
   Media Type=0x3, SPI Context=0x678D2480, Stream CallId=66
   TX Dynamic Pt=0x0, RX Dynamic Pt=0x0
*Mar 1 01:09:05.531: //65/C113522D805A/CCAPI/ccSetDigitTimeouts:
   Initial Digit Timeout=5000(ms), Inter Digit Timeout=10000(ms)
*Mar 1 01:09:05.535: //65/C113522D805A/CCAPI/ccSetDigitTimeouts:
   Call Entry(Inter Digit Timeout=10000(ms), Initial Digit Timeout=5000(ms))
*Mar 1 01:09:05.535: //65/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
   (callID=0x41, digit_event=0x1, enable=TRUE, consume=FALSE)
*Mar 1 01:09:05.535: //65/C113522D805A/CCAPI/ccCallReportDigits:
   Enabled=TRUE, Call Id=65
*Mar 1 01:09:05.535: //65/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
   (vdbPtr=0x67213130, callID=0x41, disp=-4, digit_event=0x1, enable=TRUE, consume=FALSE)
*Mar 1 01:09:05.539: //65/C113522D805A/CCAPI/cc_api_call_report_digits_done:
   Enabled=TRUE, Disposition=0xFFFFFFFC, Interface=0x67213130, Call Id=65
*Mar 1 01:09:05.543: //65/C113522D805A/CCAPI/cc_api_call_report_digits_done:
   Call Entry(Initial Digit Timeout=5000(ms), Inter Digit Timeout=10000(ms))
*Mar 1 01:09:08.723: //65/C113522D805A/CCAPI/ccDisassociateStream:
   Record Context=0x0, Stream Call Id=66, Call Id=65
*Mar 1 01:09:08.723: //65/C113522D805A/CCAPI/ccDisassociateStream:
   Call Entry(Stream Status=1, Digit Enable=TRUE)
*Mar 1 01:09:08.735: //65/C113522D805A/CCAPI/ccAssociateStream:
   Coder=5, DTMF Relay=4, Vad=0,
   Record Function=0x0, Event Queue=0x66954714, Stream Context=678B0364,
   Record Context=0x0, Stream Call Id=66, Call Id=65
*Mar 1 01:09:08.739: //65/C113522D805A/CCAPI/ccAssociateStream:
   Call Entry(Stream Status=2, Digit Enable=TRUE)
*Mar 1 01:09:08.747: //65/C113522D805A/CCAPI/cc_api_call_associated:
   Interface=0x0, CallId=65, Disposition=0
   Play Function=0x62C2DFA8, Codec=0x5, Vad=0x0
   Media Type=0x3, SPI Context=0x678D2480, Stream CallId=66
   TX Dynamic Pt=0x0, RX Dynamic Pt=0x0
R1#
R1#
*Mar 1 01:09:16.143: //65/C113522D805A/CCAPI/ccDisassociateStream:
   Record Context=0x0, Stream Call Id=66, Call Id=65
*Mar 1 01:09:16.143: //65/C113522D805A/CCAPI/ccDisassociateStream:
   Call Entry(Stream Status=1, Digit Enable=TRUE)
*Mar 1 01:09:16.175: //65/C113522D805A/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Mar 1 01:09:16.179: //65/C113522D805A/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
*Mar 1 01:09:16.227: //65/C113522D805A/CCAPI/cc_api_call_disconnect_done:
R1#Disposition=0, Interface=0x67213130, Tag=0x0, Call Id=65,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
*Mar 1 01:09:16.235: //65/C113522D805A/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*Mar 1 01:09:16.235: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 1 01:09:16.235: :cc_free_feature_vsa freeing 67445940
*Mar 1 01:09:16.235: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 1 01:09:16.235: vsacount in free is 0
I have some basic VXML gateway configuration
R1#show run
Building configuration...
Current configuration : 2588 bytes
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname R1
boot-start-marker
boot-end-marker
no aaa new-model
memory-size iomem 5
ip cef
no ip domain lookup
ip domain name lab.local
ip auth-proxy max-nodata-conns 3
ip admission max-nodata-conns 3
multilink bundle-name authenticated
voice service voip
sip
header-passing
application
service cvperror flash:cvperror.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
service new-call flash:bootstrap.vxml
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
service ringtone flash:ringtone.tcl
paramspace english language en
paramspace english index 0
paramspace english location flash
paramspace english prefix en
service ringback flash:ringtone.tcl
service handoff flash:handoff.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
service bootstrap flash:bootstrap.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
archive
log config
hidekeys
interface FastEthernet0/0
ip address 192.168.1.253 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
no ip http server
no ip http secure-server
control-plane
dial-peer voice 9191 voip
service ringtone
session protocol sipv2
incoming called-number 91919191
codec g711ulaw
no vad
dial-peer voice 9292 voip
service cvperror
session protocol sipv2
incoming called-number 92929292
codec g711ulaw
no vad
dial-peer voice 1503 voip
destination-pattern 1503
session protocol sipv2
session target ipv4:192.168.1.57
dtmf-relay rtp-nte h245-alphanumeric h245-signal
codec g711ulaw
ip qos dscp cs3 signaling
no vad
dial-peer voice 100 voip
destination-pattern 10..
session target ipv4:192.168.1.100
dial-peer voice 811 voip
service bootstrap
session protocol sipv2
session target sip-server
incoming called-number 81T
codec g711ulaw
no vad
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous
line aux 0
exec-timeout 0 0
privilege level 15
logging synchronous
line vty 0 4
login
end
Please help me out how o fix this issue/

Thanks for the respose
The microapp what I use is user.microapp.ToExtVXML[]
This is the activity log in the CVP application. I dont see any error logs.
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,newcall,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,ani,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,areacode,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,exchange,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,dnis,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,uui,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,iidigits,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.125,CVP Subdialog Start_01,enter,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.593,CVP Subdialog Start_01,exit,done
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.593,Greetings,enter,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.597,Greetings,interaction,audio_group,initial_audio_group
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.703,Greetings,exit,done
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.703,Menu,enter,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.707,Menu,interaction,audio_group,initial_audio_group
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.875,Menu,exit,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.875,,end,how,hangup
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.875,,end,result,normal
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.875,,end,duration,1
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,newcall,
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,ani,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,areacode,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,exchange,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,dnis,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,uui,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,iidigits,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,CVP Subdialog Start_01,enter,
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.921,CVP Subdialog Start_01,exit,done
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.921,Greetings,enter,
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.957,Greetings,interaction,audio_group,initial_audio_group
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:07.109,Greetings,exit,done
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:07.109,Menu,enter,
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:50:25.718,,end,how,app_session_complete
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:50:25.718,,end,result,timeout
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:50:25.718,,end,duration,1819
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,newcall,
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,ani,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,areacode,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,exchange,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,dnis,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,uui,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,iidigits,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,CVP Subdialog Start_01,enter,
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.750,CVP Subdialog Start_01,exit,done
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.750,Greetings,enter,
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.782,Greetings,interaction,audio_group,initial_audio_group
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.937,Greetings,exit,done
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.937,Menu,enter,
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:53:25.718,,end,how,app_session_complete
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:53:25.718,,end,result,timeout
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:53:25.718,,end,duration,1829
Thank you

Conference Call Single Disconnect

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