Unable to transfer outside calls twice with CCME 4.0
Hello,
I have a 3825 with a CCME 4.0 and one E1. There is an IVR (from stonevoice) implemented.
When the operator receives the call, she is able to transfer the call to a internal extension (for example 230), but when the user (ext. 230) tries to transfer the call again to another internal extension (ext. 255), the outside caller is still listening the moh and the second destination extension (ext. 255) don't receives the call.
On the phone screen of the ext.230, remains the 2 calls on hold with a message like "transfer not valid or unable to transfer".
I have enabled the h450.2 and h450.3 services:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
service script1 http://192.168.10.114/fw/Apps/IOSIVR/Server/TargetScript/script1.tcl
paramspace english index 1
param operator 99910
param msg_attesa_selection NONE
param alert_time 15
paramspace english language en
param msg_attesa_operator NONE
paramspace english location http://192.168.10.114/fw/Apps/IOSIVR/Server/AudioFiles/
param num_selection1 99906
param num_selection2 99907
param num_selection3 99908
param num_selection4 99909
param welcome_msg _menu_GC.au
param num_selection5 99910
paramspace english prefix en
service cua flash:app-b-acd-2.1.0.0.tcl
param queue-len 20
param aa-hunt1 99900
param queue-manager-debugs 1
param number-of-hunt-grps 2
param aa-hunt2 99901
service grupocastilla flash:app-b-acd-aa-2.1.0.0.tcl
paramspace english index 0
param drop-through-option 1
param second-greeting-time 30
param drop-through-prompt _dt_prompt.au
paramspace english language en
param max-time-vm-retry 2
param voice-mail 1991
param max-time-call-retry 700
param aa-pilot 99910
param number-of-hunt-grps 1
paramspace english location flash:
param handoff-string grupocastilla
param call-retry-timer 15
param service-name cua
dial-peer voice 104 voip
service grupocastilla
destination-pattern 99910
description queue loopback
session target ipv4:192.168.10.50
incoming called-number 99910
codec g711ulaw
telephony-service
load 7960-7940 P0030702T023
load 7914 S00104000100
load 7905 CP7905080001SCCP051117A
load 7920 cmterm_7920.4.0-02-00
load 7912 CP7912080001SCCP051117A
max-ephones 168
max-dn 500
ip source-address 192.168.100.1 port 2000
timeouts interdigit 5
system message CCME
cnf-file perphone
user-locale ES
network-locale ES
time-zone 26
time-format 24
date-format dd-mm-yy
voicemail 1999
max-conferences 24 gain -6
call-forward pattern .T
moh music-on-hold.au
web admin system name ird secret 5 $1$9raP$zImXvNHnQIUFoM.EfkeP31
time-webedit
transfer-system full-consult
transfer-pattern ...
transfer-pattern ....
transfer-pattern 1...
transfer-pattern 9...
transfer-pattern .T
after-hours block pattern 1 0906 7-24
after-hours block pattern 2 0905 7-24
after-hours block pattern 3 080 7-24
after-hours block pattern 4 000 7-24
create cnf-files version-stamp 7960 Sep 21 2006 18:07:37
ephone-hunt 1 longest-idle
pilot 99900
list 99980, 99981
timeout 14, 14
statistics collect
description Operator
Can you test the following and tell us what happens ?
a. Call 230 from an internal extension, transfer to internal extension B, and then do a second transfer from B to C.
b. Call 230 from outside, transfer to internal extension B , and then do a second transfer from B to C.
c. Call the operator from inside, transfer to 230, and then do a second transfer from 230 to 255.
d. Call the operator from outside, transfer to 230, then transfer to 255. (We know this fails, so this is just FYI, no need to test this).
HTH
Sankar.
Similar Messages
-
Hey,
I get this error message when calling into an unassigned number which redirects to a response group:
From user URI:
sip:[email protected];gruu;opaque=srvr:Microsoft.Rtc.Applications.Acd:RS6nRGV9DlmpNsLtmz5qeQAA
To user URI:
0220198611;phone-context=DefaultProfile
From user agent:
RTCC/4.0.0.0 Response_Group_Service Announcement_Service
Diagnostic header:
26005; reason="The Response Group application was unable to transfer the call to the configured destination and no fallback exists."
Interestingly "To user URI: 0220198611;phone-context=DefaultProfile" is the number off the caller not the destination. I wonder is this a bug? So is the response group trying to transfer to this number and failing because of course it doesnt exist?
As you can see the below the number I am calling is not 0220198611:
From phone number: 0220198611;phone-context=DefaultProfile
To phone number: +6493760053 From mediation server: onzlyncfe1.domain.co.nz To mediation server: From gateway: 192.168.100.70
To gateway:
Disconnected by: +6493760053
Does the calling party's number have to be normalised? If so how can I do this because the global normailisation rules dont seem to apply
in this situation. These rules do work when when calling into a users DDI.
Also to be clear....
+6493760053 is an unassigned number which is setup to redirect to a response group.
If I assign +6493760053 to a user then it works.
Additionally this works perfectly when the gateway sends the call to our legacy 2007r2 mediation server then on to Lync. If the gateway sends the call directly to the co-located Lync mediation server I get the error described.
I hope I make sense. If you are confused let me know :)
Help is appreciated.
Thanks,
AndrewHi ANdrew
Kindly advise how you transfered the unassigned numbers to a specific user, i used the below command but it failled, the message displayed but the call never routed:
New-CsAnnouncement -Parent service:ApplicationServer:LyncFE.squareone.local -Name "SQ unassigned number announcement" -TextToSpeechPrompt "You entered an invalid extinsion you will be forwarded to the operator" -Language "en-US" -TargetUri "sip:[email protected];user=phone"
While [email protected] is the sip uri in my lync for the operator
could you advise what is my issue? -
Bad bug: transfer outside calls to paging, holds system hostage!
I've been doing UC500's for 6 years, and I never knew about this bug until a large client of mine experienced it not once, but twice. I have recreated the problem on my UC560 as well. Here's what happens:
1. Outside call comes in.
2. Staff answers the phone, but instead of hitting the softkey "Park" they hit the speed dial "Page".
3. Staff member hits the transfer softkey
Now the outside caller is on the overhead page to all phones, with no way to reach them to reconnect or disconnect! While paging is active, staff cannot use their phones!! All handsets are tied up by the paging system. The first time this happened, the caller hung up after a minute. The second time it happened, the caller yelled to the kids in the background, said some choice curse words, and finally hung up after 2 minutes.
I recreated this problem, but have not found a way to retrieve the call and disconnect it. Any suggestions?? Any way to stop it from happening in the first place??
thanks!
tracyWell of course it depends on the client... some really do want to be able to transfer an external call to the paging. In any case, you can hang up an incoming page to "get your phone back"... though I doubt this really helps you.
The only possible workaround for your scenario that I can think of would be messing with the transfer-pattern's in CLI (under telephony-service -- see the "Cisco CME Administration Guide") to block transfers to the paging extensions.
-Dan
Please rate useful posts. -
Somehow my code is being called twice with the result that everything has the same name twice (multiple names are legal according to the DTD).
My parsing code looks like this:
public Channel getChannel(Node channelnode)
Channel channel;
NodeList channelchildren;
Element channelelement;
channelelement = (Element)channelnode;
if(channelelement.hasAttribute("id"))
channel = new Channel(channelelement.getAttribute("id"));
channelchildren = channelnode.getChildNodes();
for(int i = 0 ; i < channelchildren.getLength() ; i++)
if(channelchildren.item(i).getNodeType() == Node.ELEMENT_NODE)
if(channelchildren.item(i).getNodeName().equals(("display-name")))
channel.addDisplayname(getDisplayname(channelchildren.item(i)));
if(channelchildren.item(i).getNodeName().equals(("icon")))
if(channelchildren.item(i).getNodeName().equals(("url")))
else
channel = null;
return channel;
public String getDisplayname(Node displaynamenode)
Element displaynameelement;
Node displayname;
displaynameelement = (Element)displaynamenode;
displayname = displaynamenode.getFirstChild();
if(displaynameelement.hasAttribute("lang"))
return displayname.getNodeValue();
else
return displayname.getNodeValue();
The code above is used to parse part of this tag in my XML file:
<channel id="001.tv.tv2.dk">
<display-name>TV 2</display-name>
<icon src="http://tv.tv2.dk/images/logo/1.gif" />
</channel>
The problem is that the addDisplayname in the Channel class is somehow called twice even though there is only one <display-name> tag.
Any help figuring out what is happening will be greatly appreciated.I know about the second point you are making. The if-then-else statement is actually the start of further development since the <display-name> element can have a lang attribute and I have to deal with it, but for now I ignore it since it is extremely rarely used and other things are more important.
I had thought about the case that you mention with multiple calls of getChannel() but my debug output suggests that it is not the case. The method getChannel() returns a Channel object, which has a toString() method, and that object is given as argument to System.out.println() and never saved anywhere.
The toString() method of the Channel object looks like this:
public String toString()
String tmp = new String();
tmp += "display-name: " + displaynames.get(0);
for(String name : displaynames)
tmp += ", " + name;
return tmp;
With an output that look like this for the XML example given in my original post: display-name: TV 2, TV 2
With this output I don't see how multiple calls of getChannel() can be the cause, since multiple calls should give multiple Channel objects and multiple lines of output. -
Unable to transfer a second incoming call from my Cisco IP Phone 7940
1) A call comes into reception (call A).
2) The receptionist answers the call A / Client A.
3) While the receptionist is talking with the client (A), the screen displays the options 'Transfer' etc until then she can transfer the call.
4) Then comes a second call, the call 'B', and then 'transfer' softkey disappear displays options 'Answer' . At that time the receptionist can not transfer the call 'A' until she answers the call B.
She couldn't transfer calls until she answers all incoming callsduplicate
https://supportforums.cisco.com/discussion/12255956/unable-transfer-second-incoming-call-my-cisco-ip-phone-7940 -
CUEAC unable to transfer call.
CUEAC unable to transfer call. The following error was returned: no more Service queue devices available
Thanks for the reply tonyperla
I have 10 Service Devices lines configured. I synced with call manager several times and all the service lines are registered. -
I have updated my iphone 4 with iOS 6, but now i am unable to transfer my music videos to iphone.I cant add a music video to itunes using 'add file to library' and then transfer it to my iphone. Am using itunes 10.7.
You need to update iTunes to 11.1 on your PC
-
We are trying to 'reconnect' files. we seem incapable to do so, for some reason. Try as we might, we aren't able to do it. can anyone help?
Please see other thread
Re: We are trying to transfer files from PS Elements 13 to our web builder. We are getting a '?' in the top left hand corners of some of the image boxes. It seems that we are unable to transfer images with this ? on the image. We have spent a long time -
I have been working on initializing a some elements in the @PostConstruct annotated method within an EJB that is already annotated as @Singleton. I decided that I wanted the initialization to be done when the EJB is deployed so I added the @Startup. Both the constructor and @PostConstruct method is getting called twice. Can someone please explain to me why it appears the EJB is being initialized twice and if there is a way to prevent the @PostConstruct method from being called twice?
Many thanks,
- JoeIs this with GlassFish?
-
Since I upgraded to IOS 7.0.2, many times I am unable to answer the calls. Then I have to switch off and on to answer the calls. I checked with my three friends who updated and they are also facing same problem. Kindly let me know the solution.
Since other millions of people updated and do not have that problem - it means one of two.
1.Your friends lied to you and do not have that problem and in that case only your installation of ios7 got corrupted and you need to restore your phone.
2.Your friends didn't lie to you and all 4 of you need to restore phones (that way reinstalling ios 7 properly)
here are instructions for restore and you may have to backup before restore.
iTunes: Restoring iOS software - Support - Apple -
Hi, I am trying to Sync Iphone4 with iTunes of my Desktop, but unable to transfer my purschases to iPhone.
On Itunes i can see that Apps are added on my iPhone, but after hitting the Sync button, it tries to do the Sync & nothing happens
I Have tried this on several times but nothing happens.
Any solutions are highly appreciated.
Thanks & regards,Hey guys, I found a solution. I had the same problem, the sync will say "Determining apps to sync" in step 4 - 7 then suddenly just quits. So I did some research, apparently a lot of people have this problem, but everyone were just trying on their own PC, so I decided to try on my other laptop. My iPod synced perfectly with the other laptop so I knew it was the PC that was the problem, not my iPod. So basically it's just iTunes. I followed these steps to completely remove and reinstall iTunes on my PC and Viola! It syncs perfectly again!
Here's the link to the steps: http://www.imobie.com/support/completely-remove-and-reinstall-itunes.ht
Note: Before I did this, BeBuoi said to remove all the apps and resync, but that didn't work for me. So when I did all these steps, I had no apps in my iPod.
Goodluck! -
i am having bugs with the ios 5, after it got the update it is unable to make outgoing calls. in-place the incoming and outgoing texts are working please help me asap...!
Sometimes and this is network dependant if they suspect the phone to be lost or stolen as in this case with change of Sim card and provider then the origonal network can and some will block the phone untill you have rang them and proven it's not the case or if you have bough this 2nd hand then the origonal seller may have stopped paying the contract bill and thus the phone is blocked
-
Unable to perform call transfer or call park for an outbound call via SIP Trunk (SKYPE)
We have configured the SIP Trunk & SIP profile and successfull make outbound call through SIP Trunk (SKYPE). However, we are not able to perform call transfer or call park when the call is connected.
The scenario is:
A call to an phone number via SIP trunk, when call established, A perform call-transfer to B. After the call-transfer, the call Drop and Phone B show error code "Temp Fail"
When i select "enable MTP" in SIP trunk, we are able to call transfer and call park. But it limit the number of call session to 1.You are probably running into some sort of Codec issue. IE, your phone is G.711 and the trunk is G.729. You will need to transcode the call at somepoint.
-
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 -
Trouble with CCME 4 and VIC2-2FXO; IOS 12.4(9)T
Trouble with CCME 4 and VIC2-2FXO; IOS 12.4(9)T
I am having trouble making outgoing call or answering incoming call.
When I try to call out from my IP 7961 phone, it fails with the message "unknown number".
For incoming call, it rings but when I pick up the call nothing happens,
Put the receiver back on hook, the phone carries on ringing. I am in UK
and just trying to set up test system with one analogue line. Any help will
be most appreciated. My config of the 2811 router is posted below. All calls ineternally works fine.
Thank you for your help.
hostname Test-CME
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 10.10.10.1 10.10.10.10
ip dhcp excluded-address 10.139.139.1 10.139.139.10
ip dhcp pool host
network 10.10.10.0 255.255.255.0
default-router 10.10.10.1
option 150 ip 10.10.10.1
ip dhcp pool data
network 10.139.139.0 255.255.255.0
default-router 10.139.139.1
dns-server 10.139.139.5
voice-card 0
no dspfarm
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
h323
sip
header-passing
registrar server expires max 3600 min 3600
interface FastEthernet0/1
no ip address
no ip mroute-cache
duplex auto
speed auto
no shut
interface FastEthernet0/1.2
description ** Data VLAN **
encapsulation dot1Q 2
ip address 10.139.139.1 255.255.255.0
interface FastEthernet0/1.3
description ** Voice VLAN **
encapsulation dot1Q 3
ip address 10.10.10.1 255.255.255.0
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
tftp-server flash:S00104000100.sbn
tftp-server flash:TERM41.7-0-3-0S.loads
tftp-server flash:term61.default.loads
tftp-server flash:term41.default.loads
tftp-server flash:CVM41.2-0-2-26.sbn
tftp-server flash:cnu41.2-7-6-26.sbn
tftp-server flash:Jar41.2-9-2-26.sbn
tftp-server flash:term70.default.loads
tftp-server flash:term71.default.loads
tftp-server flash:cnu70.2-7-6-26.sbn
tftp-server flash:Jar70.2-9-2-26.sbn
tftp-server flash:TERM70.7-0-3-0S.loads
tftp-server flash:CVM70.2-0-2-26.sbn
control-plane
voice-port 0/3/0
connection plar opx 202
caller-id enable
dial-peer voice 1 pots
incoming called-number .
destination-pattern 9T
port 0/3/0
telephony-service
load 7914 S00104000100
load 7941 TERM41.7-0-3-0S
load 7961 TERM41.7-0-3-0S
load 7970 TERM70.7-0-3-0S
max-ephones 20
max-dn 40
ip source-address 10.10.10.1 port 2000
calling-number initiator
service phone videoCapability 1
system message MKC CME
url services http://10.10.10.1/voiceview/common/login.do
url authentication
http://10.10.10.1/voiceview/authentication/authenticate.do
time-zone 21
date-format dd-mm-yy
voicemail 600
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
moh music-on-hold.au
web admin system name admin secret 0 test
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 9.T
secondary-dialtone 9
create cnf-files
ephone-dn 1 dual-line
number 201
label 201
description Sarah
name Sarah
ephone-dn 2 dual-line
number 202
label 202
description Vitthal
name User2 Vitthal
ephone-dn 3 dual-line
number 203 secondary
label 203
description Neil
name User3 Neil
ephone 1
video
username "user1" password 201
mac-address 0018.18EE.947F
type 7961 addon 1 7914
button 1:1
ephone 2
video
username "user2" password 202
mac-address 0018.18BB.B973
type 7941
button 1:2
ephone 3
video
username "user3" password 203
mac-address 0018.1885.6BA2
type 7970
button 1:3Hi
Please find enclosed debug attachment for voice ccapi and ephone. First, I called from outside. Extension 202 rings but when I answered on extension 202 nothing happens. Replace the rceiever and the pone starts ringing again.Second step. I tried to call out by dialing 9 and then number but after a while phone displays unknown number.
Thank you for your help.
Vitthal
Maybe you are looking for
-
I am trying to upgrade to iTunes 11.4 on my Windows7 PC and get an error message that says " iTunes has an invalid signature. It will not be installed" How do I fix that?
-
Multiple pages in sap=script
HI , I have a scenario in which the header window has to be displayed in all the consecutive pages.In the layout i gave consecutive pages and created header window in each page. But the data in the header window of the first page is not displayed in
-
HT201077 How do I change the primary email linked to my Photo Stream
As I no longer use this primary email I need to be able to put in my new gmail email.
-
Texfield autoSize problem inside embedded movie
Hi guys! I have a strange problem.. I have a flash file inside which i have a movieclip. I am loading an external swf file into this movieclip using the loadMovie script. the external swf has a couple of texfields that load in content from an XML fil
-
Hey, Apple, fix podcasts!
I notice many people are having the same problem I am with podcasts not showing up on their nanos in iTunes. I too would like to delete some podcasts from my nano, but they are nowhere to be found when my ipod is connected to my computer through iTun