Transfer calls from Unity Connection doesn´t work

[aherraez77]

Hello:

I have the following issue:

For historical reasons when internal users wants to call the operator they call the main number of the office and with a translation pattern they are redirected to the Unity Connection. They listen to the greeting and if they don´t enter any extension the transfer rule connects them to the operator.

This works only for one site while when the call is from the second site, after the greeting the users don´t  connect with the operator and the call is disconnected.

The users of the second site don´t have problems to call the operator directly, only fails when the call is transfered from the Unity Connection.

Can anyone give some help? I don´t know what can be going.

Best Regards.

[ramans2]

Hi,

step 1   Please check when users from second site calls which call handler/user they hit.

To determine that you can use port status monitor tool.

step 2. Once you know where the call hits go to that call handler/user click edit »» greetings»»standard greeting

Check what is set as after greeting option.

Step 3. If it is set to hang up or change it and send it to call handler operator, choose transfer to operator than send directly to greeting.

Step 4  Go to that operator>> edit >> transfer rule>> standard put the operator call handler.

If this operator is allready been used by other site.

Create a dummy operator call handler and choose that in step 3 and in new operator call handler send the call to second site operator extension.as step 4

Thanks

Raman

[aherraez77]

Hi Raman:

The second site hit the same call handler than the first site. I have only one call handler.

I have tried to use the operator handler and after the greeting transfer to that call handler. The result is the same as the call goes to that handler, the user listen the message "you are been transfered" and then the call fails.

How can I see what happen after the call is trying to be transfered to the final extensión of the operator?

Best Regards.

[nipuri]

so if i am understanding it correctly 

Both the sites hit the same call handler  and the then after greeting option has operator in int which transfer the call to operator fir site 1 but it fails for site 2.

I have couple of questions

so we have both sites congigured on same call manager ?

Do we have diffrent operators for both sites ?

is it a sip or skinny integration between call manager and unity ?

we can track call in call manager traces or CDR.

Thanks and Regards

Raman

[aherraez77]

Hi Nipuri:

Both sites are configured on the same call manager.

The operator is the same for both and the integration is skinny.

I attach the sdl trace files to add it to the other files I have attached.

Best Regards

[Ayodeji Okanlawon]

What are the call details please. How do we know what to look for in your cucm logs..

Calling number:

called number:

xfered to number:

time of call:

[aherraez77]

Hi Ayodeji,

the calling number is 245.

the called number, after a translation pattern is 8000 (the UC). In the transfer configuration is the xfered number 860.

The 860 extension has configured a Forward no answer to the 299. 

the call was originated at 12:41:27

Although in this call the call goes to the 299 instead the 860 the failed happens in both situations. 

Best Regards

[Ayodeji Okanlawon]

Ok here is my analysis..

INVITE from 245 to 0917020274.xlated to 8000 then goes to vmpilot 8200..
CUCM selects vmport 101, then transfer to 860 with cfa 299.

Now 299 has a cfna to 220 with a cfna timer 10s

cfna    = 220, cfnaToVM    = 0, cfnaCss = pt_internal:pt-pstn, cfnaTimer = 10

+++ So here is what is happening +++

At 12:43:20.729, CUCM selects ANN to play ring back for the final transferred party 299 and 245. Here is CUCM selecting ANN to play ring back

10146169.001 |12:43:21.173 |AppInfo  |AnnDControl(ANN_2) - star_StationStartMediaTransmissionAck - No Agena PID exists for PartyId=16783327, CI=24468649

++++ Next the call rings for a while and and no one answered the call, CUCM then disconnects the ANN because it needs to perform the cfna ++++

10146451.001 |12:43:30.801 |AppInfo  |ARBTRY-ConnectionManager- wait_MediaDisconnectRequest CI(24468636,24468649

+++ Now pay attention to the timeline...its exactly 10sec now, which is your cfna timer on the extension 299 +++

++ Here we see CUCM disconnecting both parties (245 and 299 from the ANN, to stop ring back playing) +++

10146491.001 |12:43:30.919 |AppInfo  |ARBTRY-ConnectionManager- wait_MediaDisconnectReply CI(24468636,24468649),disconnectType(0),IFHandling(1,0)
10146496.001 |12:43:30.920 |AppInfo  |AnnDControl::sendPolicyAndCACUnregisterRequest- ANN CI = 24468647

+++ Next CUCM does Digit Analysis to send the call to 220 +++

10146502.006 |12:43:30.921 |AppInfo  |Digit analysis: match(pi="1", fqcn="245", cn="245",plv="5", pss="pt_internal:pt-pstn", TodFilteredPss="pt_internal:pt-pstn", dd="220",dac="0")
10146502.007 |12:43:30.921 |AppInfo  |Digit analysis: analysis results
10146502.008 |12:43:30.921 |AppInfo  ||PretransformCallingPartyNumber=245
|CallingPartyNumber=245
|DialingPartition=pt_internal
|DialingPattern=220
|FullyQualifiedCalledPartyNumber=220
|DialingPatternRegularExpression=(220)

++++ Now it looks like 220 is set to forward the call to VM for some reasons ++

++Looks like 104 is one of your VM ports +++

10146508.002 |12:43:30.922 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 192.168.14.49 on port 49481 index 25 with 1569 bytes:
[5118603,NET]
INVITE sip:104@192.168.14.10:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.14.49:49481;branch=z9hG4bK0f65dd01
From: <sip:299@cucm>;tag=00082f1b433cbc7959c7a4b9-5c97edd1
To: "VoiceMail" <sip:104@192.168.14.10>;tag=2025483~171ffb25-f099-f151-f142-7979a73e03e4-24468633
Call-ID: ad080680-665170ca-2a19-a0ea8c0@192.168.14.10

As you can see the call is almost going into a loop.

Suggestions

1. Increase the CFNA on extension 299 from 10 to something like 20 to give operator time to answer the call

2. Review your whole setup. This solution looks very messy..

3. Tell the operator to answer the call on time!!!

[Dennis Mink]

If your call handler and  transfer are the same for site 1 and 2, one owrks the other doesnt, to me this sounds like the issue is most likely related to Call Manager. I am thinking maybe codec (region) or calling search space.

Can you provide us will call traces for a fialed call to see how call manager deals with the transfered call?

here is how: http://ciscoshizzle.blogspot.com/2015/05/using-rtmt-to-trace-calls.html

[aherraez77]

Hi Dennis:

I attach you the call flow and the details of the message 15 -> Update. 

As you can see the set  up is ok, and the destination answer de call but the call doesn´t have any sound.

Best Regards

[Jaime Valencia]

How many VM ports do you have configured for outbound dialing??, and what's the busy trigger on the operator's line??

[afsal abdul gafoor]

* Please check the advertised codec settings in unity connection,might be the codec used for site to site call is not advertised in unity connection.

[aherraez77]

Hi Afsal,

the codec between both sites is G729 and it is advertised in UC. Apparently the call is anwered, after the tranfer, but there is any sound

Best regards

[Carl Ratcliffe]

Can you confirm the Unity Connection ports have a CSS that can that can reach the extension of site 2 phones.

Thanks, Carl Ratcliffe

Preston Lancashire England

[aherraez77]

Hi Carl,

The UC ports have a css that can reach the extensión of site 2. 

Best Regards