Cisco CUCM 9.1 Call Queuing no Ring Back

I have a customer that has a hunt group that has 6 users in it, they are routing the calls by longest idle. The way the customer handles the inbound call is with the operator-> then they transfer the call to the hunt group. They want to be able to present MOH when they transfer to the hunt group. I enabled Call Queuing for this hunt group and set the MOH source to the customers recording , and work great as long as all agents are busy. When the callers are trasfered to the hunt and agents are available they hear the ringback tones as the call goes from agent to agnet until it gets answered. The customer is wanting to eliminate the ringback tone during the hunt cycle and just play MOH. I know this is possible with UCCX, but they are not willing to purchase UCCX. Is there any way i can silence the ringback tone and have MOH while the call is hunting?

Hi, there is a product intended exactly to do that:
http://www.imagicle.com/go/queuemanager
It extends the embedded call queuing capabilities and is already certified with 9.1, available in solution catalog and also providing advanced historical reporting.
Licensed per channels and not per operator/agent, cheap, dramatically EASY to deploy, config and manage.
Offer a free of charge supervisor app on iPAD and can be combined with the imagicle operator console (http://www.imagicle.com/go/bluesattendant) for a full, professional, out of the box customer service solution.
You can contact imagicle for more info or download the free 30 days eval.
Regards.
Christian Bongiovanni
coCEO and CTO
Imagicle SpA

Similar Messages

Use Cisco CUCM for outbound "call me at" feature on Lync meetings

I'm trying to find a step by step to enable users (non enterprise voice users) to use the dial me at feature in Lync conference meetings. I only want the user to have the ability to tell Lync to dial a number to place that number into the conference call,
the feature is easy to enable but i can't get the routing right between CUCM and Lync. I've looked all around the net but I can't seem to find anything that matches what i'm trying to do, other docs cover enterprise voice and that's out of my scope. Any assistance
here would be nice. Thanks

Hi,
In Lync Server 2010, it is not supported with “call me at” function for non-Enterprise Voice users.
However, Lync Server 2013 now allows participants that are not Enterprise Voice enabled to initiate dial-out calls from a meeting conference, called “Dial-out Conferencing for non-Enterprise Voice users”.
This can be configured by setting the Conferencing policy to allow this feature (Set-CSConferencingPolicy –AllowNonEnterpriseVoiceUsersToDialOut:$true). After enabling this, then assign a voice policy to the users who need the function.
Best Regards,
Eason Huang
Please remember to mark the replies as answers if they help, and unmark the answers if they provide no help. If you have feedback for TechNet Support, contact [email protected]
Eason Huang
TechNet Community Support

CUCM 9.1 native call queuing - calls stuck in queue

Hello everyone
We are using the CUCM 9.1 native call queuing feature for attendants hunt group an have the following problem - if all the attendants are busy, the next arriving call gets into queue, receiving MOH treatment, as described in documentation. But if one or more attendants gets free and ready to receive calls, queued call doesn't get transfered to his/her phone. Logging in additional attendants, logging them in and out again doesn't change anything, once the call got in queue it stays in queue.
Exact cucm release version is 9.1.1.20000-5.
Has anyone seen this? Any ideas on what traces/logs to check?

Hi,
it seems you hitting this bug CSCuc16486
http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCuc16486&from=summary
HTH
Anas
please don't forget to rate the helpful posts

Caller doesn't hear a ring back if the call is ringing on agent phone

Hi team,
can anybody help me? Here in Germany there are a lot of customers that don't understand why the caller doesn't hear a ring back when the call is ringing on agent phone for instance with UCCX 8. I read some discussions here for that issue but the only solution for that should be to change the MOH on CUCM. This is not a very good solution when the call in queue is set to hold he will hear the ring back tone in the queue as well :-(
Has anybody another solution for that issue???
All the discussion I've read was for IPCC version 4 or so. Is there another solution for this issue in version 8.x
In one discussion I found the following solution:
Yes, To resolve this issue,Set up a Send H225 User Info Message service parameter for Cisco CallManager service in CallManager. Perform the following steps:
In Cisco CallManager Administration, select Service > Service Parameter.
Select the correct server from the drop-down list.
Set the service to Cisco CallManager.
In the Send H225 User Info Message field, under the Cluster Wide Parameters (Device - H323) section, select H225 Info for Ring Back.
Reset the H.323 voice gateway.
After completing this procedure, the caller hears ring back when the agent phone is ringing.
Does it realy work?
Need you help!
Thanks in advance.
Tobias

Hi Tobais
you right one of the solutions is to change the MOH to the ring back one
however there are two types of MOH that you need to select
use MOH is the music to be played while the call on hold/ in queue
use Network MOH which is the oneyou need to change to the ring back MOH source file so when the call get transferred to an agent ring back MOH will be played
these options under the CTI ports in UCCX
also you can search the forum here and there are many discussions about this topic
HTH
if helpful Rate

CUCM 8.6 Call Forwarding to External Number Issue

Hello,
Call forwarding worked without problems, we could forward our phones to external numbers and everything was ok, when somebody called to my phone, I could got the call to my cell phone.
But now when I forward my phone to external number and try to call to my phone I get busy trigger.
We didn't change configuration or install any update.
I think its my ISP-s problem, to whom we have SIP Trunk.
I don't understand log file, so can you tell what is the problem?
Here is log:
057729XXXX is called party, cell phone number
original calling party number is 240XXXXX, but it is forwarded to 2484XXX
INVITE sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191815-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5060>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052091 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29790 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 2|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpHandler::fireSignal - SEND: index = 2, handler = 0xb2d59c98|*^*^*
17:34:18.526 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP-IP:5060|*^*^*
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1172, ISP-IP:5060)|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
17:34:18.536 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 358 from ISP-IP:[5060]:
[12623361,NET]
SIP/2.0 100 Trying
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/ccsip_spi_get_msg_type returned: 2 for event 1|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/context=(nil)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/gConnTab=0xf484290, addr=ISP-IP, port=5060, connid=2, transport=UDP|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Return existing connection for port 5060 connId 2|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Checking Invite Dialog|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0xb1b50c90/INVITE response with no RSEQ - disable IS_REL1XX|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_TRYING value=500 retries=3|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/States/0xb1b50c90/0xb1b50c90 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.561 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.561 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
17:34:18.561 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 396 from ISP-IP:[5060]:
[12623362,NET]
SIP/2.0 403 Forbidden
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
Contact: <sip:ISP-IP:5060>
[12623363,NET]
ACK sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0
INVITE sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191816-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5062>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052092 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29792 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xa6b4d7c0|*^*^*
17:34:18.567 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP's-Other-IP:5062|*^*^*
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1177, ISP's-Other-IP:5062)|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input - handle = 335|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 0|*^*^*
17:34:18.569 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 394 from ISP's-Other-IP:[5062]:
[12623365,NET]
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0
17:34:18.587 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 375 from ISP's-Other-IP:[5062]:
[12623366,NET]
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
CSeq: 101 INVITE
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0
|2,100,230,1.4901099^ISP's-Other-IP^*
[12623367,NET]
ACK sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0

SIP/2.0 403 Forbidden error
If your router is sending a SIP/2.0 403 Forbidden error to the SIP server you are registered to, there is a good chance your router is blocking the incoming call due to the toll-faud prevention feature that was added to IOS version 15.1(2)T.
How to Identify if TOLLFRAUD_APP is Blocking Your Call
If the TOLLFRAUD_APP is rejecting the call, it generates a Q.850       disconnect cause value of 21, which represents ‘Call Rejected’. The       debug voip ccapi inout command can be run to       identify the cause value.
Additionally, voice iec syslog can be       enabled to further verify if the call failure is a result of the toll-fraud       prevention. This configuration, which is often handy to troubleshoot the origin       of failure from a gateway perspective, will print out that the call is being       rejected due to toll call fraud. The CCAPI and Voice IEC output is demonstrated       in this debug output:
%VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected):
IEC=1.1.228.3.31.0 on callID 3 GUID=F146D6B0539C11DF800CA596C4C2D7EF
000183: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallSetContext:
   Context=0x49EC9978
000184: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 3 with tag 1002 to app "_ManagedAppProcess_TOLLFRAUD_APP"
000185: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallDisconnect:
   Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
The Q.850 disconnect value that is returned for blocked calls can also       be changed from the default of 21 with this command:
voice service voip
ip address trusted call-block cause
How to Return to Pre-15.1(2)T Behavior
Source IP Address Trust List
There are three ways to return to the previous behavior of voice       gateways before this trusted address toll-fraud prevention feature was       implemented. All of these configurations require that you are already running       15.1(2)T in order for you to make the configuration change.
Explicitly enable those source IP addresses from which you would like           to add to the trusted list for legitimate VoIP calls. Up to 100 entries can be           defined. This below configuration accepts calls from those host           203.0.113.100/32, as well as from the network 192.0.2.0/24. Call setups from           all other hosts are rejected. This is the recommended method from a voice           security perspective.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
Configure the router to accept incoming call setups from all source           IP addresses.
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
Disable the toll-fraud prevention application completely.
voice service voip
no ip address trusted authenticate
Two-Stage Dialing
If two-stage dialing is required, the following can be configured to       return behavior to match previous releases.
For inbound ISDN calls:
voice service pots
no direct-inward-dial isdn
For inbound FXO calls:
voice-port
secondary dialtone

Decode Error in Cisco VCS for a call from Jabber to telyHD

Hi,
We have the following setup:
Cisco Jabber <---> Cisco VCS <---> telyHD (Our Video Product)
Now, the call works well when the call is intiated from telyHD to Cisco Jabber.
But, the call fails when the Cisco Jabber initiates the call towards telyHD.
The packet capture on VCS shows that there is a syslog message from VCS to Jabber indicating the following when 180 Ringing is received from telyHD:
Message [truncated]: tvcs: Event="Decode Error" Service="SIP" Src-ip="10.1.1.19" Src-port="5060" Dst-ip="10.1.11.11" Dst-port="25331" Detail="(User-Agent) Single header User-Agent appeared twice in message" Protocol="TCP" Level="1" UTCTime
Can you please help in troubleshooting this issue. I don't see multiple "User-Agent" headers in the 180 Ringing sent by telyHD.
I have attached the packet capture on VCS.

I am also having the same issue. Can someone please respond? Thanks.

BACD and hold times for Call Queue

Hi Everyone,
Is it possible to include hold times/wait times for BACD call queues? Also, anyone run into issues when configuring BACD and the time it takes for it to ring in? We had an issue previously that where the call would enter the queue, wait 30 seconds or so, and then ring out. Unless this is normal.
Thanks as always,
HelpDesk

Hi Help Desk,
Its been a while since I configured this and I am not sure if the current versions of CCA allow for this change to take place.
I know on the command line that this is possible, but ultimately it would be better to use CCA to make the changes as I am not sure if it would recognize the change if you did it via CLI.
In the past as an alternative to hold times I did some work around's by pushing the BACD to an AA and putting in there some rudimentary options that would allow a user to go back into the line, but they would loose their position in doing so, but this shouldn't be a big problem.
Also in terms of entering the BACD, I havent see the 30 second delay before so I am not sure what is going on there and I cannot say for that that it is normal operations for it.
Cheers,
David Trad.
davidtradconsultinggmail.com

BE6000 Native Call Queuing Design Constraints?

We see the doc wiki has not been updated with any UCM 9.0 native call queuing design constraints. We would assume that the call queuing at a min would need to meet the MoH design constraint of 50 sessions, since you figure there is a high probability that one call will be queued in each of 50 queues with MoH playing with each.
http://docwiki.cisco.com/wiki/Supported_System_Capacities
Is there any other performance impact we should consider or can we ask the doc wiki be updated with design constraints around call queuing?
Can the BU clarify is that is 50 simultanous MoH streams per cluster, or 50 per server? We can eliminate the load by going to multicast and IOS distributed MoH for remote sites, but something tells me there probably is a max queued call ceiling we need to consider for this platform.

At the risk of getting caught up in the pass-by-reference/pass-by-value debate :), simple values (ints chars etc). are passed as values, and modifying them on the C++ side of the procedure call will not modify the values on th java side.
Possible solutions are:
1)pass an array of (say) ints. Then, modifying individual values should modify the java-side values (much in the way that the Stream.read(byte[]) functions do.
2)format up a strin g with your return values, return the string and parse it on the java side.
3)have fields in the java side that are accessed and modified by your native call.
4)pass an object with some fields in it. Have the C++ access and modify the object's fields.
Hope this helps.

Cisco CUCM 6.1.5 CodeYellowExit alarm(s) received

Hi,
I am running Cisco CUCM ver.6.1.5 & two CUCMs acting as Publisher & Subscriber, Total Phones registered on CUCM is 94.
The Problem that I am facing i used to receive the below Alarm & I noticed the CPU is hitting 85% on both CUCMs so I found the Process called CCM is utilizing 80% of CPU usage.
From Tue Feb 23 23:38:05 AST 2010 to Tue Feb 23 23:59:05 AST 2010 on node 172.30.145.4, there are 1 CodeYellowEntry alarm(s) and 1 CodeYellowExit alarm(s) received. On Tue Feb 23 23:58:35 AST 2010, the last CodeYellowEntry alarm generated: CodeYellowEntry AverageDelay : 31 EntryLatency : 20 ExitLatency : 8 SampleSize : 10 TotalCodeYellowEntry : 26 HighPriorityQueueDepth : 0 NodeID : CUCM-A
Note:
I had disabled all Traces for all CUCM types but still the problem exist.
I had enough MEM & CPU.
Please Advice !!!
Best Regards,
Mohanad

Did you recreate your whole cluster, just as it is in production, on UCS???
I did the procedure from 6.1(5) and 7.1(5) and had no problems.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

Call cancelled while waiting 180 ring back

I have a cisco as5400 as voip gateway, when call coming in from PSTN T1, the gateway send invite to our sip server, the sip server send back 100 trying, but didn't send back 180 trying within about 10 seconds, the cisco send cancel to sip server to end the call, how can i change the cisco configuration to let wait cisco wait longer for the 180 ring back?

Hi, yes there are timers and retries that can be set, see below. However i'm not sure if that would fix it, check also you q.931 debugs, it is possible the disconnects comes from telco and not the gateway, 10 seconds while waiting 180 is not excessive.
sip-ua
timers ?
buffer-invite Time to buffer the INVITE while waiting for display info
connect Time to wait for confirmation a session connected
connection Connection related timers
disconnect Time to wait for confirmation a session disconnected
expires Time to wait for the expiration of an INVITE request
hold Time to wait during hold before disconnecting
info Time to wait before INFO retransmission
keepalive Options keepalive related timers
notify Time to wait before NOTIFY retransmission
prack Time to wait before starting PRACK retransmission
refer Time to wait before REFER retransmission
register Time to wait before REGISTER retransmission
rel1xx Time to wait before starting reliable 1xx retransmission
trying Time to wait for provisional response to INVITE
update Time to wait before starting UPDATE retransmission
retry ?
bye BYE retry value
cancel CANCEL retry value
info INFO retry value
invite INVITE retry value
keepalive KEEPALIVE retry value
notify NOTIFY retry value
prack PRACK retry value
refer REFER retry value
register REGISTER retry value
rel1xx Reliable 1xx response retry value
response Response Methods retry value
subscribe SUBSCRIBE retry value
update UPDATE retry value

PSTN Caller no ring back tone when IPPhone is CallForwarded

CallManager version 4.1(3)SR with 6608 T1 blade. IPPhone is call forwarded to an external PSTN number. PSTN caller dials IPPHone DID number, Phone is forwarded to external number via 6608 T1 PRI, PSTN user hears no ring back. User most of the time hangs up. Any help would be great.

Hi John,
You could try changing this setting on the Callmanager Gateway config page (after hours);
Setup non-ISDN Progress Indicator IE Enable
Default leaves this setting disabled (unchecked).
Enable this setting only if users are not receiving ringback tones on outbound calls.
When this setting is enabled, the Cisco CallManager sends Q.931 Setup messages out digital (that is, non-H.323) gateways with the Progress Indicator field set to non-ISDN.
This message notifies the destination device that the Cisco CallManager gateway is non-ISDN and that the destination device should play in-band ringback.
From this doc;
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803ed699.html#wp1311922
Hope this helps!
Rob
Please remember to rate helpful posts...........

"Call Queue List" Idea

To the great folks of Apple:
We've all had the experience of using cell phones on the go and not wanting to have to pull over to look up numbers. Sometimes, before I jump in my car, I'll call the 3 or 4 numbers I intend to call one by one, then hang up before it rings. This way the numbers I intend to call are in my "recent calls" list. Why not create a Call Queue List that you could prepare and have them ready to call with the least hassle while you're on the go?
Jim Wilson

http://www.apple.com/feedback/
Regards.

Hello i have a unusual problem,anyway my iphone 4 wont turn on and when someone calls they hear ringing,but my phone doesent ring,so i open cimcas now ring off. Screen is just black,because it's turned off..Please help me to solve this problem..Thanks

Hello i have a unusual problem,anyway my iphone 4 wont turn on and when someone calls they hear ringing,but my phone doesent ring.So i poen simcase and now ring off. Screen is just black,because it's turned off..Please help me to solve this problem..Thanks

As far as trying to power up your device, make sure it has a good charge and then hold the button on the top of the phone and the home button on the faceplate together until the apple appears on the screen.

Lync 2013 not hear ring back in mobile when call by number

Hi,
I have setup a Lync server 2013 standard Front end on windows server 2012 R2 and also installed update for Mobility service and enabled it. It seems everything works perfect but...
When I call by number in android phones and iphones, I do not hear ring back tone, although call is proceeded and if the calling person answer the call, everything will be fine. The strange thing is that if I call by name I hear ring back
tone.
This problem does not exist while using windows phone or desktop client.
Thanks, Amir.

Early media refers to media (e.g., audio and video) that is exchanged before a particular session is accepted by the called user.
If the remote party does not generate it, then you will not hear it. As stated by Holger, the logs will have this information. Without the logs it is not possible to identify where the problem is. Your app might have received it and not played it or it was
never send.
Regards Herbert Zimbizi

All data and messaging works on my note 4, but last night the cellular phone service just went dead. I can call people but the line is just silent, and people can call me, it rings, but I can't hear them. ?????

all data and messaging works on my note 4, but last night the cellular phone service just went dead. I can call people but the line is just silent, and people can call me, it rings, but I can't hear them. ????? What is wrong with it???? How do I fix it????

You aren't attached to a bluetooth headset or other device for some strange reason? Or even an automobile bluetooth, I have a friend that has this issue with his Prius because even when it is not running the bluetooth is still active and connects to his phone.

Cisco CUCM 9.1 Call Queuing no Ring Back

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