CUIC Reporting - Calls Transferred

Hey Guys,
I need to create a report to show how many calls were transferred from an agent, and to what destinations.
I've read the CUIC doesnt tap into the right database to collect this type of information, is this correct?
If anyone could shed some light on how to configure another database source or where to go to configure such a report..
Thanks
Ben

Bump?
I know it's an old thread, but I've got the same issue.
UCCX 9
CUCM 9
CSQ calls get transferred by an agent to a non-agent destination and I'm being asked to provide reports on how many calls were transferred to which extension, and preferably including which agent transferred the call.
-Mike

Similar Messages

UCCX 9.0(2) - CUIC Reports Call Abandoned calls inconsistencies

Hi,
We are finding that the there are inconsistencies in the abandons figures shown for a CSQ report for 'Contact Service Queue Activity Report' and the calls that are shown in the 'Abandoned Call Detail Activity Report'.
For example if I view a CSQ in 'Contact Service Queue Activity Report' I see that there are 18 calls abandon today with 9 dequeued. However I when run 'Abandoned Call Detail Activity Report', I can only see 4 calls showing for that CSQ. Before I got to TAC with this has anyone else seen this?

Hello Saeed, Kiarash,
Page 212 on this document explains why.
http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_7_0/programmin...
Q.Why do the number of abandoned calls shown on the Abandoned Call Detail Activity Report and the number of calls shown on the Contact Service Queue Activity Report not match?
A.These values can differ because a Contact Queue Detail record may mark a call as dequeued while a Contact Call Detail record marks the call as abandoned. For example, consider the following workflow:
Start
Accept
Prompt
Select Resource
-Connect
-Queue
--Play Prompt (Prompt2)
--Dequeue
--Play Prompt (Prompt3)
End
If a call is abandoned during Prompt2 or Prompt3, the Contact Queue Detail record will mark the call as dequeued but not abandoned from any queue. The Contact Call Detail record will mark the call abandoned.
In this case, the Contact Service Queue Activity Report will show the call as dequeued from all CSQs for which the call was routed. The Abandoned Call Detail Activity Report will show the call as abandoned from all from all CSQs for which the call was routed.
Another reason is that the call gets aborted after being in the queue.
Such calls get marked as aborted in the Contact Call Detail record, and as dequeued (if it was dequeued before aborting)/abandoned in Contact Queue Detail record. In this case the CSQ reports will display more abandoned calls than the Abandoned Call Detail Activity Report.
Thank you,
Shadi

CUIC report for the Call transferred.

Hello,
I am looking for CUIC report for the number of calls transfered by an agent. Below is the query that works fine in SQL, but i am unable to generate a CUIC report on this.
select AgentPeripheralNumber,count(*) from t_Termination_Call_Detail where
where DateTime between '02 Aug 2013 00:00' and '18 Sep 2013 23:59'
and
ICRCallKey in (select ICRCallKeyParent from t_Termination_Call_Detail
where DateTime between '02 Aug 2013 00:00' and '18 Sep 2013 23:59'
and CED='2345')
group by AgentPeripheralNumber
Regards,
Sandeep

Try querying "Termination_Call_Detail" instead of "t_Termination_Call_Detail".
Check out the CallDetailReport.xml report available here:
http://developer.cisco.com/web/ccr/documentation
That should be a good starting point for you.
-Jameson

CUIC reporting: handled calls greater than answered calls

Hello,
We met the following problem in Cisco Cuic reporting 8.5.4 for cisco UCCE 8.5.3.
We use only the stock reports. In call type historical all fields we met a strange situation for a simple inbound script:
- the numbers of handled calls is greater than the number of answered calls. This is vey unusual because any call
is first answered and only after that is handled or not by the agent. So the handled condition supose to be first answered.
Any ideas? What could be the cause for additional handled calls?
Also we have problems between the number of handled calls in Agent Team historical all fields and the number of handled calls
in Call type historical all fields. Should not be the same number in both reports ?
The definitions for answered, handled, offered are very simple and they lack essential info in describing the situations
they treat. The whole documentation for enterprise misses many times essential info and the only source of real
info is others knowledge and experience.
Any help is wellcome.
Best regards,
Marius

Answered is incremented when the Agent picks up the call (or received the activity, in the case of EIM).
Handled is incremented when the call (or EIM activity) completes.
Handled will be incremented regardless of whether the call is transferred/conferenced/consulted or not.
As others have mentioned, Answered and Handled counts can be incremented during different intervals depending on the length of the call. The only way I can think of where that would be the case here would be if your agents are handling EIM activities... They could have 5 open emails in their inbox at the end of the day - these 5 open emails would be considered Answered, but not Handled.
If this isn't from non-voice activities, then perhaps you have some system issue causing the problem.
-Jameson

Report on transferred calls

I need to report on transferred calls. Agents are using standard CAD and selecting the transfer icon. Within the transfer form, they use the softphone to dial various 800 numbers. I need to create a report that I can group by 800 numbers called using the CAD transfer option. Does this come from the call manager? We recently implemented CVP. Are there any CVP reports that would provide this information?

Hi Ryan
I'm afraid there isn't one - only the CSQ Activity reports and the CSQ Service Level Priority Summary reports seem to show the service level info, and those are summary reports rather than reports that show call lists.
You would need to get a custom report put together for this... it's an Informix SQL procedure-writing and Crystal Reports exercise to do that.
Regards
Aaron

How can I use either the contact center reporting or CAR to identify calls transferred from agents to other extensions?

We are not to the call manager and I am trying to find a way to report calls that were received to our customer service agents though the call queue but were subsequently transferred to an extension out side of the queue. With our implementation we changed call routing and need to see how many misrouted calls are getting to the queue.

Marc,
Are you on UCCE or UCCX? If you're using UCCE, the Termination_Call_Detail table includes a record of all transfers made by Agents.
-Jameson

CUIC reporting in UCCX 9

Dear Experts,
I have 2 issues with CUIC reporting.
the first issue is related to filter in the report. for example, for agent detail summary, I want to apply 2 filter, the first one is the agent name, the 2nd one is the call type (incoming, outgoing, ..etc ). it only apply the agent name and forget the call type. it apply only one filter.
how we can apply 2 filters or more ?
the 2nd issue is related to the custom report, I want to create a custom. the documentation said, you need to right click on the category or sub category and create a new one. unfortunately, I use the right click and nothing happened.
waiting your feedback.
Thanks in advance
Anas

Amer is referring to CUIC Premium, not CCX Premium. CUIC is a separate product also used with UCCE and comes in two editions. Cisco has packaged the CUIC Standard edition with CCX as it is primarly attempting to be an HRC replacement first. HRC only allowed one filter criteria as well.
CUIC Premium is not cheap for production use (just wait until you ask for a quote)! In the second situation the suggested workaround is to buy the non-production lab license of CUIC Premium, design the report definition there, then export and import it into the embedded CUIC instance. Don't forget you would also need to build a new stored procedure in Informix to go with the report. Alternatively, you could subcontract the report development to a partner who already has this in place.
Please remember to rate helpful responses and identify helpful or correct answers.

Call transferring- Not transferring calls

I use call tranferring alot but my skype does not tranfer calls to my local mobile number.
Whenever skype is active, i get calls on skype. I started using the call transferring feature so that i can have my skype calls transferred on my local mobile numbers when internet connection is not available.
But my calls are not been tranferred now.
Why is this happening?

Hi,
well, Tanner has a point, it must be solved processwise.
The company I used to work for had a customer with a large UCCE contact center and they asked for a report that would show this kind of suspicious activity. UCCE has actually a way to mark these calls, unfortunately, as far as I know, there is no such thing in UCCX. The rule was simple: you can be caught only once doing that.
G.

CUIC Report - Queue Status

Dear Group
I am recently been started working with UCCE platform, for one of our client who is using CTI Desktop Client (V8.5) required to have
A field in the CTI screen about the number of calls currently queued or waiting in the queue
A CUIC report for the manager to show the status of the queue or no of calls waiting in the queue waiting to allocated to an agent
Please let me know the best way to achieve this
thanks in advance
Rajesh

The agents can click on the stats button of their CTIOS desktop and will see how many calls are in queue for the skill groups they are in. For CUIC, run a call type report and report on the call type right before the call enters the queue. This will give you the best and most accurate information about the queue.
david

Packaged CCE 9.03 - Display Reason Code Text in CUIC report

The (Stock) Agent Team Real Time report includes a Reason Code field, but not the descriptive Reason Code Text (or Name). Is there any way to present the descriptive text or name in the report?
Please note that we have CUIC standard.
Thanks!
Jackie

Comparing Handled on an Agent report to Answered on a Call Type report is tricky, since depending on your call flows the two values can be vastly different. If you're looking at Answered and Handled on the same report (Call Type Historical All Fields should have both), there are opportunities for either value to be higher than the other within any given interval or across multiple intervals.
Generally, if Answered is higher than Handled in one interval, then Handled is going to be higher than Answered in one of the intervals after that (though not necessarily the one right after - depends on call length). In the database, Answered is incremented in the interval when the call is answered, Handled is incremented in the interval when the wrap-up completes. So a very basic example would be with a single call - Say the Agent picks up an inbound call at 8:56am, and hangs up and completes wrap-up at 9:07am... The 8:30am interval would show 1 call Answered and 0 Handled, while the 9:00am interval would show 0 calls Answered and 1 Handled.
-Jameson

[UCCX] Create/Edit CUIC Report Definitions

Is the ability to create/edit cuic report definitions available with Enhanced or is that Premium only?

I found this - https://supportforums.cisco.com/thread/2222847
If I'm reading this correctly (which I very well might not be), it requires premium licensing. Maybe it'd be better for me to explain what I'm after since I might not even be on the right track.
I've created copies of a couple of stock inbound reports and am trying to add grouping by state (via calling party number npa/nxx). 99% of the inbound calls are apparently from 3 or 4 states, so I figured it'd be pretty easy to implement but I haven't come up with anything yet. Any ideas would be much appreciated.
thanks,
will

CUIC, Handled Calls and Handled Time problem

Hello,
Our costumer uses Outboud Predictive Dialer, UCCE. Everthing works fine but in CUIC reports we do not have almost any information about Handled calls and Handled Time (example in attach).
Any hints are much appreciated!
Dusan

Hi,
alright.
One day you will outgrow CUIC anyway, so if I were you I would start thinking about creating a custom reporting solution (if you are a Microsoft shop, it's SRSS, if Java, then perhaps BIRT).
G.

4.5 to 6i migration or report calling from forms in XE???

Hi Friends,
I really appreciate your helping tendency. Let me explain the whole scenario.
My application basically consists of
Forms [32 Bit] Version 5.0.6.8.0 (Production)
Report Builder 3.0.5.8.0
Oracle Enterprise Edition Release 9.2.0.1.0 - Production
Now we migrated our database into Oracle Database 10g Express Edition Release 10.2.0.1.0 - Production. After the migration all the existing forms and report individually were running without any problem. But there was a huge problem. Whenever I tried to call any report from any form the report engine appeared and then stopped responding. So report calling from forms was not possible after the database migration. I was suggested by someone from OTN forum to migrate my forms and reports into 6i. I have started doing the migration using ifcmp60. WIth forms 6i, reports can be called from forms. But I am facing problem in every other steps. There are many things are not working after the migration though the migration log shows no error. For example, COMMIT_FORM, EXECUTE_QUERY (for detail table), LOV, TOOL.GETVAR lots of things are not working.
My application has 200 forms and 250 reports.
Please help and advise on:
How to call reports from form when the database is XE Or
How to overcome the stated migration problem
Rgds,
Luther

Dear Mark:
I have two choices. I can either migrate to 6i or I can go with the existing 4.5. If I go with 4.5 ......the reports are not being called from forms. This problem is happening after I changed the database from 9i to XE. Please advise what to do so that I can keep my forms in 4.5, database in XE and can call reports from forms.
Rgds,
Luther

Oracle reports calling problem from FORMS

Hies
Guys I've a problem in oracle report calling from Oracle FORMS. I am using reports and forms 6i. When the report name length is more than 30 characters the report engine runs first time correctly but on the second call of report the system gives an error :
" RWRBE60.exe has generated errors and will be closed by windows.You will need to restart the program ".
After this error the report engine closes and i have to call the report again.
Is there a length fixed for reports name in Oracle 6i.
Is any possible solution available.
Remember i dont want to shorten my report names.
Kindly help me with this.
Thanx.

Thanx Frank
Well one solution is that i can decrease the length of my report name and the problem does not occur than.
any ways thanx ..but where to get the patch from.
Kindly provide me with the link if possible.
Thanx,
Qaiser Qayyum Malik.
[email protected]

CUCM 8.6 Dropped call transfers involving SIP phones

Hi All,
I am a developer who has been tasked with figuring out why call transfers are being dropped by Cisco CUCM when the original call comes from a SIP phone. This scenario works:
Cisco phone calls another Cisco phone, which transfers the original call to a SIP phone
These scenarios do not work:
SIP phone calls Cisco phone, which transfers the original call to another Cisco phone
SIP phone calls Cisco phone, which transfers the original call to another SIP phone
I have researched the Call Manager traces to the best of my ability, and I see some info in there that could potentially point to the source of the problem. I am just unable to understand what the trace means:
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/active_CcDisconnReq: ccDisconnReq.onBehalfOf=Media : ccDisconnReq.s.sv=2 : ccDisconnReq.c.cv=47 |1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/Stack/Info/0x0/sipConstructContainerContext #### Created container=0xb0b42f58|1,100,71,1.1^*^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|1,100,63,1.93259^10.10.10.85^*
I have been wondering whether this could be a codec issue, however the SIP phones we are using are configured with the following codecs:
G711U
G711A
G722
ILBC
GSM
and our SIP software is also set to accept the first codec offered by the remote side. It seems from the SIP client logs that G722 is being used as the codec to communicate with the Cisco phones, but perhaps I'm misinterpreting.
I have attached a CUCM trace of a call from a SIP phone (ext. 491) to a Cisco handset (ext. 170) where the Cisco handset attempts to transfer the call to another SIP phone (ext. 492). The trace snippet shown above is from this log.
I would really appreciate it if someone more experienced with VoIP/SIP/CUCM could take a look and offer any ideas on what the issue might be, and also how we might be able to address it. I can try to provide more info about our CUCM configuration if needed.
Thanks in advance!

Leslie, so here is what I found from the traces....
To understand the difference we need to understand how cucm performs call transfers from a sccp signalling point and a sip signalling point
SCCP
When the transfer key is pressed
1. CUCM sends a CloseReceiveChannel and StopMediaTransmission to the IP phone involved in active media (referenced by the callids)
NB, here CUCM updates the call state on the phone to a call state of 8 which is "Hold"
2.CUCM tells the held party to listen MOH from MOH server
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5..CUCM sends a CloseReceiveChannel between the held phone and MOH server (to tear down the media)
6. Next CUCM sends a CloseReceiveChannel and StopMediaTransmission to the transfering party & transfered party to remove Xferring party from call
7. finally CUCM sends OpenReceiveChannel between the original called party and the transfered party..and call is done
For SIP signalling. when the first transfer key is pressed
1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
2. CUCM sends a Delayed offer to insert MOH or to resume Media stream
NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
6.Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call
Now having explained all of these, we need to look at where the call is failing for SIP-----SCCP----SIP calls without MTP
lets look at succesful SCCP-----SCCP-----SIP without MTP
Point 4 above
++++++++Extension 170 presses the transfer button to connect the two calls (Callid=24378483)+++++++++++++
(0003395) SoftKeyEvent softKeyEvent=4(Trnsfer) lineInstance=1 callReference=24378483
Point 5 above
++++Next CUCM closed the media between extension 160 and MOH server callid=24378480(this is the only active call on this callid)+++
(0003396) CloseReceiveChannel conferenceID=24378480 passThruPartyID=16777845. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
Point 6 Above
+++++Next cucm closes the call between extension 170 and 490 callid=(24378483)++++++++
(0003395) CloseReceiveChannel conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
(0003395) StopMediaTransmission conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
Point 6 above for the sip side (since the destination is SIP, to tear down media to SCCP phone, so as to connect the caller to the xfered party)
+++++++Next CUCM sends a re-invite with a=inactive SDP to the sip phone ++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885626,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23332dbee978
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
o=CiscoSystemsCCM-SIP 192115 2 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0
m=audio 24560 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=inactive-----------------------------------------------------Inactive
Still part of Point 6 for SIP signalling
++++++++++++Next sip phone responds with a 200 OK recevonly SDP +++++++++++++++++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885628,NET]
SIP/2.0 200 OK
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
a=ptime:20
a=recvonly-------------------------------------a=recvonly
Finally Point 7 above..
+++++++++++++=Next cucm sends a DO re-invite to extension 492-sip phone++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885630,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
+++++++Next we get a 200 OK from sip phone with sdp=sendrecv+++++++++=
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885634,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
Contact:
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
Call-ID: [email protected]
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
m=audio 16574 RTP/AVP 9 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
+Now CUCM sends an OpenReceiveChannel and start media xmission to sccp phone (callid=24378480) with media parameters of sip phone++++++
(0003396) OpenReceiveChannel conferenceID=24378480 passThruPartyID=16777848 millisecondPacketSize=20 compressionType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierIn=? sourceIpAddr=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104). myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
(0003396) startMediaTransmission conferenceID=24378480 passThruPartyID=16777848 remoteIpAddress=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104)
remotePortNumber=16574 milliSecondPacketSize=20 compressType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierOut=?. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
+++++++++++=Next Phone sends its ACK+++++++++++++++
(0003396) OpenReceiveChannelAck Status=0, IpAddr=IpAddr.type:0 ipAddr:0x0a0a0a89000000000000000000000000(10.10.10.137), Port=20352, PartyID=16777848
+++++++++++=Next CUCM sends ACK to 200 OK from SIP Phone+++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885635,NET]
ACK sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23366067b8c0
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
Date: Tue, 19 Feb 2013 21:44:45 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.137
b=TIAS:64000
b=AS:64
t=0 0
m=audio 20352 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Now at this point all is well...and the call is connected....
Now here is where the call is failing on the SIP-SCCP-SIP call without MTP
From Point 2 above, CUCM sends a DO to insert MOH, and then gets response, then sends an ACK to 200 Ok back to SIP Phone..
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881160,NET]
ACK sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22035ecc1fcb
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:38:50 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Max-Forwards: 70
CSeq: 102 ACK
o=CiscoSystemsCCM-SIP 190666 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------IP address of MOH server
t=0 0
m=audio 4000 RTP/AVP 0--------------------------------MOH port 4000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendonly---------------------------------------------------------sendonly
+++++NOW Point 6 above (SIP) CUCM sends a=inactive to break media path to MOH server to connect caller and xfered party++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881161,NET]
INVITE sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:39:04 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
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: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 164
v=0
o=CiscoSystemsCCM-SIP 190666 4 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
t=0 0
m=audio 4000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=inactive---------------------------------------------------------------------media inactive
At this point, we should get a response back from the sip phone...
and here is what we got..
++Trying which is expected++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 331 bytes:
[881162,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 103 INVITE
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Content-Length: 0
++++++++Then we get a BYE+++++++++++++++
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 576 bytes:
[881163,NET]
BYE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.104:53361;branch=z9hG4bKa2vdQvR7J9OiMyjU;rport
Contact:
Max-Forwards: 70
From: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.8
To: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 3 BYE
Content-Length: 0
So this is the root cause of the problem. Your SIP phone does not know how to respond to multiple media break between it and the MOH server.
The difference between this and the succesful SCCP-SIP--SCCP, is that the held party was a sccp phone, hence the sip phone only has to process one a=inactive SDP message, where as in the SIP-SCCP-SIP, the help party was sip, so the sip phone has to process two a=inactive SDP messages
Now what is the difference when MTP is involved! A Big difference. MTP stays in the media path. So there is never a break in media and no inactive SDP attribute is sent. The flow looks like below:
for the initial call...The SIP phone sends its media to MTP and likewise the SCCP phone
SIP------Media------MTP------------Media-------SCCP Phone
When the new destination is dialled and transfer is commited,
SIP-------------media----MTP--------media---------MTP
The final invoite sent to connect 492 and 491 has MTP as the IP address to connect Media to.
++++++++Ivite to 492 ++++++++++++++
INVITE sip:[email protected]:61303;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231a3b24b862
From: ;tag=192048~d8e94532-127d-4dca-bba0-64b1675da032-24378472
To: ;tag=78FF5BF6C019A55EA020B69BB6A767E2
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: [email protected]
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: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 214
v=0
o=CiscoSystemsCCM-SIP 192048 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------------------------------the MTP ip address
t=0 0
m=audio 25038 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++Invite to 491 +++++++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.94 on port 50376 index 1887
[885429,NET]
INVITE sip:[email protected]:50376;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231b78d1b56
From: ;tag=192046~d8e94532-127d-4dca-bba0-64b1675da032-24378467
To: "491" ;tag=F13CE94DE942C47680356A647DC7F916
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: AE7045FFB2D8D9C28D54651473A14A5D41B5B93C
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
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: "Leslie2" ;party=calling;screen=no;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192046 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195----------------------------------------MTP
b=TIAS:64000
b=AS:64
t=0 0
m=audio 25030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Wao! That was a long one isnt it...It was fun too.
So now you can look at your sip phones and see if they can accept two inactive sdp messages within the same call. That way you can remove MTP. otherwise you will have MTP involved in every single call involving a sip phone, even if they do not involve transfers
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