mae45674: SIP REFER issue in CUBE against Avaya

[mae45674-paetec]

Hello Guys:

Greetings. I have an issue with a CUBE with Incoming calls to the IPPBX (Avaya).  The call comes from the SP to the IPPBX via CUBE and then the user will transfer the call to a third number. As soon as the third number rings, the transfer button on the phone is pushing and the call drops. I tried different configuration to add SIP REFER in the CUBE but still not success. Any thoughts? 

I changed the following in the CUBE for SIP REFER:

1) clock timezone gmt 0  ==> didn't work

2) voice service voip  ==> didn't work

   no supplementary-service sip refer

   supplementary-service media-renegotiate

3) Remove the parameters on 2) and  add the following to dial-peers  ==> didn't work

   dial-peer voice 1 voip   ( and dial-peer voice 5001 voip)  => Facing to/from SP/CUBE

     voice-class sip asserted-id pai
     voice-class sip referto-passing

Thank you

Martin. 

[Rolando Valenzuela]

Can you confirm the call flow?

This is what I'm understanding:

------- ITSP -> CUBE -> Avaya -> Phone [1] -------
At this point a person answers the phone, it is a working call and then this call is forwarded to phone #2?

If that is correct, where is phone 2? it is also controlled by avaya?
under a different scenario transfer calls works between phone #1 and phone 2?

Thanks!

Rolando A. Valenzuela.

[mae45674-paetec]

Rolando:

Greetings. The phone 2 could be any number, but not Avaya DID. Could be PSTN, Cell, etc.. 

A = Avaya number

B = PSTN 

C = PSTN

B calls A and A click transfer button and dial to C. As soon as C start ringing, A hangup up the call or click on transfer button. Normal conditions, B and C will talk each other. In this case, call drops. 

Let me know if it is better explanation now. :) 

Thank you

[Rolando Valenzuela]

I'm still kinda confused, B and C will always talk to each other they are outside your organization, do you meant A and C?

Can A and C talk normally?, the problem only happens when do do a transfer B -> A -> C?

Do you need CUBE in order to complete the call A -> C?

Is there any codec/protocol change during the process?

You will have to check each leg of the call, a signaling flow may help you to find the route case.

https://andrewjprokop.wordpress.com/2014/03/17/refer-revisited/

Rolando A. Valenzuela.

[mae45674-paetec]

Rolando:

Greetings. Yes, it is the call flow that I am talking about. I need to know where the REFER will be config in the CUBE to complete the call flow. In conclusion, B and C needs to be talk together via A (Avaya phone).

Thank you.

Martin. 

[Rolando Valenzuela]

You don't configure a REFER message, and you definitely do not want the Avaya phone as an intermediary.

You haven't answer any of the questions about where CUBE sits for this transfer, codec situation and others so I will suggest to run "debug ccsip messages" and your CUBE and determine who sends the BYE or any other disconnection message and go from there.

Check this: https://community.cisco.com/t5/collaboration-voice-and-video/sip-debugging-commands-overview/ta-p/3155271

Rolando A. Valenzuela.

[mae45674-paetec]

Rolando:

Good morning. I have the following in the CUBE:

Dial-peer 5101 (from PBX to CUBE)

Dial-peer 5100 (from CUBE to PBX)

Dial-peer 1 (from CUBE to SP)

Dial-peer 5001 (from SP to CUBE)

Incoming call from SP to PBX-Phone is taking DPs 5001 and 5100. Call is completed and two way audio. Then PBX-Phone will transfer the call to a third party number. It uses DP 5101 and 1, then when the x-transfer button is pressed, the call drops.

The BYE came from the CUBE outside IP to PBX; however, before the BYE, PBX sent the REFER message to CUBE outside IP and I got a Port Unreachable message (ICMP) between the REFER and BYE message from the PBX to CUBE-Outside.

In other words, the REFER message is not passing the CUBE inside to SP, it keeps on the PBX side. 

PBX   --> 5101    <=> x.x.x.x  <=>  y.y.y.y  --> 1             <=>  SP

         <-- 5100           (out)              (in)     <-- 5001

I hope that now my explanation has been better.  The CUBE outside ip is facing to PBX (x.x.x.x); meanwhile, CUBE in ip is facing to SP (y.y.y.y).

Thank you.

Martin.

[Rolando Valenzuela]

Do you still have that capture? the REFER (packet 506) should give you the new IP/PORT to where send the call to.

After a REFER the call will be tear-down between CUBE and PBX since it is CUBE who will route the call next.

But now everything makes more sense, have you checked this document: https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_sip/configuration/xe-3s/cube-sip-xe-3s-book/voi-cub-sip-dyn-refer-handling.pdf

Please pay attention to the restrictions, this one sounds interesting "The Dynamic REFER Handling on the Cisco UBE feature  supports only Session Initiation Protocol (SIP)-to-SIP call transfers"

Rolando A. Valenzuela

[mae45674-paetec]

Rolando:

The document that you sent me it was configured three days ago with all combinations, even REFER dynamic,

By the way, I am sending  you a copy of the SIP REFER, I hope that you can see something.

Thank you very much for your great support and I am sorry to bother you too much.

Martin.

[mae45674-paetec]

Also Rolando, if A calls B and then A put on hold and call C, and then transfer the call, B and C talk each other. Works just with outgoing calls from Avaya.

Phone (Avaya) -> CUBE -> PSTN-1

                                     |-> PSTN-2

  => PSTN-1 and PSTN-2 are connected. 

The problem here is the incoming call scenario from PSTN network. 

PSTN-1 -> CUBE -> Avaya -> Phone

                                   |-> PSTN-2 

                                   | Avaya-Phone transfer and call drops

Thank you.

Martin.