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Transfer Call To Avaya Ipo


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#1 ahmed.muneer 17 October 2017 - 10:45 PM

Dear VG Team,

Am trying to tranfer a call to our Avaya IPO by using Hookflash monitored option and mentioning the desired Avaya extension in phone field but it fails to do so.

kindly check the VGS and logs in the attachment.

kindly note am using  the HMP VOIP version 7



#2 SupportTeam 18 October 2017 - 06:31 AM

When doing a SIP REFER Transfer (RFC 3515 transfer) you need ot specify the full IP address of the extension/device to which you would like to transfer to.

 

eg. if you want to transfer to extension 112 and the IP address of the Avaya IPO is 192.168.5.1 then the transfer destination would be:

 

112@192.168.5.1

 

But this will only work if the VoiceGuide line was registered as a SIP extension (or a number of extensions) on the Avaya IP Office.

 

If the connection from Avaya IPO to VoiceGuide is done over a SIP Trunk (which is usually best approach, especially if larger number of lines are used) then the SIP REFER transfer will not work.

Avaya IPO does not support SIP REFER transfers for calls made over SIP Trunks.

 

You will need to use a 2-line "Dial-and-Conference" type transfer instead.

(a 'Dialer' license is required to make these 2-line "Dial-and-Conference" transfers)

 

 

Some more information:

https://support.avay...ead.php?p=26050



#3 ahmed.muneer 18 October 2017 - 04:50 PM

Dear Vg Team.

thanks for your reply,

already i have 4 lines of Enterprise edition, so does it support  "Dial-and-Conference"



#4 SupportTeam 18 October 2017 - 04:56 PM

A 'Dialer' license is required to perform the 2-line "Dial-and-Conference" transfer.

 

To perform the 2-line "Dial-and-Conference" transfer you will need to upgrade the "Enterprise" license to a "Enterprise+Dialer" license.



#5 ahmed.muneer 18 October 2017 - 08:08 PM

Then for this license how many lines in a time can the IVR transfer ?



#6 SupportTeam 18 October 2017 - 08:21 PM

Each '2-line' transfer uses 2 lines for the duration of the transferred call.

 

So a 4 line license can do up to 2 of '2-line' transfers at the same time.



#7 ahmed.muneer 22 October 2017 - 08:06 PM

Dears,

 

i have created a SIP extension for the IVR on the Avaya IPO and used the PBX Hookflash Transfer - Monitored , but it's not working.

Please refer to the attached screen shot.



#8 SupportTeam 22 October 2017 - 08:48 PM

No screenshot or traces were attached. Can you please try attaching them again?



#9 ahmed.muneer 22 October 2017 - 10:01 PM

kindly find in the attachment , the logs the VGS design.

  • Pbx_Hookflash Transfer - Monitored.png


#10 SupportTeam 23 October 2017 - 06:11 AM

Could you please use WireShark to capture the SIP messaging between VoiceGuide and Avaya IP Office.

 

Start WireShark (https://www.wireshark.org) capture before starting the VoiceGuide service - to have WireShark capture the extension registration as well - and then make the call into the system and have VoiceGuide perform the transfer attempt.

 

Then post the WireShark's .pcapng file of SIP messages along with the VoiceGuide's ktTel trace.

 

To view/save just the SIP messages specify this in the WireShark's 'Filter' options text box:

 

sip



#11 ahmed.muneer 23 October 2017 - 04:42 PM

So if i use 3 way conference option or  dial out and conference option it will require a dealer license ?



#12 SupportTeam 23 October 2017 - 04:49 PM

"Dial and Conference" requires the Dialer license.

 

"3 way conference" is only supported on some analog systems. VoIP does not natively support "3 way conference".



#13 ahmed.muneer 23 October 2017 - 05:24 PM

kindly find the attached required tracing files.



#14 SupportTeam 23 October 2017 - 05:35 PM

Looks like WireShark trace was not started before the VoiceGudie service was started. The SIP extension registration was not captured by WireShark

 

Also, the ktTel trace attached shows that the SIP registration was not performed successfully:

439 094633.426  4028     fn    VoIPProvider_AuthenticationSet
440 094633.429  4028           TelDriver_VoIPProvider_AuthenticationSet start. No Authorization entries defined. Exiting Authorization phase.

441 094633.432  4028     fn    VoIPProvider_Register(protocol=SIP, reg_server=192.168.5.204, reg_client=GeideaVc, local_alias=, sH323SupportedPrefixes=)
442 094633.435  4028           TelDriver_VoIPProvider_Register IP_PROTOCOL_SIP reg=[server:192.168.5.204, client:GeideaVc, alias:, expires=0, ] iptB1=1

480 094634.135  5920     ev    GCEV_SERVICERESP (board device)
481 094634.153  5920           GCEV_SERVICERESP ResultInfo: gcValue=1283(0x503|GCRV_PROTOCOL|event caused by protocol error) gcMsg=[Event caused by protocol error] ccLibId=8 ccLibName=[GC_H3R_LIB] ccValue=[0x66||] ccMsg=[IPEC_REG_FAIL_invalidAlias] additionalinfo=[]


#15 ahmed.muneer 23 October 2017 - 07:32 PM

Dear , 

 

is the below SIP extension is correct ?

 

<VoIP_Registrations>
 
<VoIP_Registration>
<Display>GeideaIVR</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.5.204</RegServer>
<RegClient>GeideaVc</RegClient>
<LocalAlias></LocalAlias>
<Expires></Expires>
</VoIP_Registration>
 
</VoIP_Registrations>
 
<VoIP_Authentications>
 
<VoIP_Authentication>
<Display></Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername></AuthUsername>
<AuthPassword></AuthPassword>
</VoIP_Authentication>
 
</VoIP_Authentications>
 
 
<Example>
 
<VoIP_Registrations>
<VoIP_Registration>
  <Display>IVRSIP</Display>
  <Protocol>SIP</Protocol>
  <RegServer>192.168.5.1</RegServer>
  <RegClient>199</RegClient>
  <LocalAlias>sip:199@192.168.5.1:5060</LocalAlias>
</VoIP_Registration>
</VoIP_Registrations>
<VoIP_Authentications>
<VoIP_Authentication>
  <Display></Display>
  <Realm></Realm>
  <Identity>199</Identity>
  <AuthUsername>199</AuthUsername>
  <AuthPassword>123456</AuthPassword>
</VoIP_Authentication>
</VoIP_Authentications>
 
</Example>
 
</VoIP_Lines>


#16 SupportTeam 24 October 2017 - 12:18 AM

In above Config.xml excerpt the VoIP_Authentication section does not have any information in it.

 

Please note that the anything between the <Example> ... </Example> tags is not read in by VoiceGuide. It is there in Config.xml as example only.

 

It looks like the 'Example' entry was changed instead of the actual entries used by VoiceGuide

 

Also, the <Identity> field should usually be left blank.

 

Please see: http://www.voiceguid...ip_register.htm



#17 ahmed.muneer 24 October 2017 - 12:26 AM

i have used the same but still not working, please find the below config file VOIP registration and is it ok to register more then one 

 

<VoIP_Registrations>
 
<VoIP_Registration>
<Display>GeideaIVR</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.5.204</RegServer>
<RegClient></RegClient>
<LocalAlias></LocalAlias>
<Expires></Expires>
</VoIP_Registration>
 
</VoIP_Registrations>
 
<VoIP_Authentications>
 
<VoIP_Authentication>
<Display></Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername></AuthUsername>
<AuthPassword></AuthPassword>
</VoIP_Authentication>
 
</VoIP_Authentications>
 
<VoIP_Registrations>
 
<VoIP_Registration>
<Display>Avaya IVR</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.5.1</RegServer>
<RegClient>199@192.168.5.1</RegClient>
<LocalAlias>199@192.168.5.204</LocalAlias>
<Expires>120</Expires>
</VoIP_Registration>
 
</VoIP_Registrations>
 
<VoIP_Authentications>
 
<VoIP_Authentication>
<Display></Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>199</AuthUsername>
<AuthPassword>123456</AuthPassword>
</VoIP_Authentication>
 
</VoIP_Authentications>
 
 
</VoIP_Lines>


#18 SupportTeam 24 October 2017 - 12:33 AM

Config.xml should have one <VoIP_Registrations> section.

 

The <VoIP_Registrations> section can have multiple <VoIP_Registration> sections in it.

 

and:

 

Config.xml should have one <VoIP_Authentications> section.

 

The <VoIP_Authentications> section can have multiple <VoIP_Authentication> sections in it.

 

 

Recommend you first begin with registering one extension, and then, once that works, add more extensions (by adding more <VoIP_Registration> and <VoIP_Authentication> sections).

 

 

If you have problems with registering please make the WireShark capture as advised before and post it along with the ktTel trace.



#19 ahmed.muneer 24 October 2017 - 01:01 AM

should i register the IVR server it self in the below way then register a SIP extension , please refer to the below configuration.

IVR IP : 192.168.5.204

Avaya IP : 192.168.5.1

 

<VoIP_Registration>
<Display>GeideaIVR</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.5.204</RegServer>
<RegClient></RegClient>
<LocalAlias></LocalAlias>
<Expires></Expires>
</VoIP_Registration>
<!-- Registering the SIP extension -->
<VoIP_Registration>
<Display>Avaya IVR</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.5.1</RegServer>
<RegClient>199@192.168.5.1</RegClient>
<LocalAlias>sip:199@192.168.5.204:5060</LocalAlias>
<Expires>120</Expires>
</VoIP_Registration>
<!-- End of Registering the SIP extension -->
</VoIP_Registrations>
 
<VoIP_Authentications>
 
<VoIP_Authentication>
<Display></Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>199</AuthUsername>
<AuthPassword>123456</AuthPassword>
</VoIP_Authentication>
<!-- Registering the SIP extension -->
<VoIP_Authentication>
<Display></Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>199</AuthUsername>
<AuthPassword>123456</AuthPassword>
</VoIP_Authentication>
<!-- End of Registering the SIP extension -->
 
</VoIP_Authentications>
 
</VoIP_Lines>


#20 SupportTeam 24 October 2017 - 05:59 AM

One <VoIP_Registration> section and one <VoIP_Authentication> section is needed for each extension that you register.

So something like this:

 

<VoIP_Registrations>

<VoIP_Registration>
<Display>Avaya IVR</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.5.1</RegServer>
<RegClient>199@192.168.5.1</RegClient>
<LocalAlias>sip:199@192.168.5.204:5060</LocalAlias>
<Expires>120</Expires>
</VoIP_Registration>
</VoIP_Registrations>
 
<VoIP_Authentications>
<VoIP_Authentication>
<Display></Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>199</AuthUsername>
<AuthPassword>123456</AuthPassword>
</VoIP_Authentication>
</VoIP_Authentications>


#21 ahmed.muneer 24 October 2017 - 05:22 PM

Dear,

i have used the same but it's not registering, please find the attached Wireshark tracing and the logs files.

Note that i have started the wire shard before starting the voiceguide.

IVR SERVER IP : 192.168.5.204

Avaya SERVER IP : 192.168.5.1



#22 SupportTeam 24 October 2017 - 06:52 PM

Looks like the WireShark trace was not started before the VoiceGuide service was started.

 

WireShark trace does not contain any REGISTER messages. (see attached screenshot of the posted WireShark trace)

 

Please make WireShark capture again, this time make sure to start WireShark capture before starting the VoiceGuide service.

 

 

Also, ktTel trace shows the SIP Registration was not successful.

435 095930.055  3712     fn    VoIPProvider_AuthenticationSet
436 095930.059  3712           TelDriver_VoIPProvider_AuthenticationSet start. adding 1 auth entries.
437 095930.059  3712           adding to auth parmblk:   199 ******

441 095930.063  3712     fn    VoIPProvider_Register(protocol=SIP, reg_server=192.168.5.1, reg_client=199@192.168.5.1, local_alias=sip:199@192.168.5.204:5060, sH323SupportedPrefixes=)
442 095930.067  3712           TelDriver_VoIPProvider_Register IP_PROTOCOL_SIP reg=[server:192.168.5.1, client:199@192.168.5.1, alias:sip:199@192.168.5.204:5060, expires=120, ] iptB1=1

452 095930.080  3712     fn    VoIPProvider_Register(protocol=SIP, reg_server=192.168.5.204, reg_client=192.168.5.204, local_alias=192.168.5.204, sH323SupportedPrefixes=)
453 095930.080  3712           TelDriver_VoIPProvider_Register IP_PROTOCOL_SIP reg=[server:192.168.5.204, client:192.168.5.204, alias:192.168.5.204, expires=120, ] iptB1=1


491 095930.140  2972     ev    GCEV_SERVICERESP (board device)
492 095930.159  2972           GCEV_SERVICERESP ResultInfo: gcValue=1286(0x506|GCRV_CCLIBSPECIFIC|event caused by cclib specific failure) gcMsg=[Event caused by call control library specific failure] ccLibId=8 ccLibName=[GC_H3R_LIB] ccValue=[0x151d||] ccMsg=[IPEC_SIPReasonStatus405MethodNotAllowed] additionalinfo=[]
493 095930.159  2972           ccValue!=IPERR_OK (VoIPProvider_AuthenticationSet)
494 095930.159  2972   1       ev idx=18 : evttype=870(870)=2160(2160) metaevent.crn=0, data=005E4B40(06DC0F60), len=8(8) q: 0/4
495 095930.159  2972     ev    GCEV_SERVICERESP (board device)
496 095930.159  2972           GCEV_SERVICERESP ResultInfo: gcValue=1286(0x506|GCRV_CCLIBSPECIFIC|event caused by cclib specific failure) gcMsg=[Event caused by call control library specific failure] ccLibId=8 ccLibName=[GC_H3R_LIB] ccValue=[0x157d||] ccMsg=[IPEC_SIPReasonStatus501NotImplemented] additionalinfo=[]
  • ws2.png


#23 ahmed.muneer 24 October 2017 - 07:55 PM

is the configuration in the previous post in the Config.xml correct ?



#24 SupportTeam 24 October 2017 - 09:15 PM

The Config.xml has two <VoIP_Registration> entries and two <VoIP_Authentication> entries.

 

Please only keep the first <VoIP_Registration> entry and the first <VoIP_Authentication> entry, then start the WireShark capture and the start the VoiceGuide service.



#25 ahmed.muneer 24 October 2017 - 10:50 PM

Dear,

 

i have followed your instruction , but same issue i think.



#26 SupportTeam 25 October 2017 - 07:10 AM

The posted WireShark trace was not started before the VoiceGuide service. Registration as an extension attempt was not captured in that trace. Please see attached screenshot.

  • ws.png


#27 ahmed.muneer 25 October 2017 - 04:22 PM

but i have started the WireShark before starting VoiceGuide service.



#28 SupportTeam 25 October 2017 - 08:40 PM

Looks like the capturing of the posted trace was not done over the relevant time as neither the initial registrations, and neither the call itself was captured.

 

Recommend you try doing the capture again and this time look at WireShark screen to make sure it is actually capturing data before the VoiceGuide service is started.



#29 ahmed.muneer 26 October 2017 - 12:05 AM

Would you please provide me with a trial license for the Dialer so i can try it first.



#30 SupportTeam 26 October 2017 - 05:44 AM

To test Dialer just rename the USERINFO file in VoiceGuide directory to something else.

 

After renaming the USERINFO file then stop and start the VoiceGuide service.

 

Make sure you wait until VoiceGuide service has fully sopped before starting it again.

 

VoiceGuide will then start in Evaluation mode, which has all features of the "Enterprise+Dialer" license available.



#31 ahmed.muneer 26 October 2017 - 04:39 PM

How to ensure it's turned into  Evaluation mode.



#32 SupportTeam 26 October 2017 - 04:46 PM

If the USERINFO file is renamed then VoiceGuide will automatically start in Evaluation mode.

 

Nothing else needs to be done.

 

The Line Status Monitor will show if software is running in Evaluation mode.



#33 ahmed.muneer 26 October 2017 - 04:56 PM

yes it's turned on , thank you very much.



#34 ahmed.muneer 26 October 2017 - 05:14 PM

it's working thank you very much for that , but i noticed it's using two lines current one that's having the call and another one for transfer.

so if i bought 2 Dialer i will be having additional 4 lines, please advise.



#35 SupportTeam 26 October 2017 - 05:23 PM

it's using two lines current one that's having the call and another one for transfer.

The "Dial and Conference" transfer uses 2 channels for duration of call.

 

 Note that this allows you to have caller continue through IVR script after the transfer connection completes, (eg. to do a port-call survey or transfer to another extension etc.) IVR monitors key-presses during the transfer as well, and can record the connected call etc.

 

If you want to use the Dialer then you will need to upgrade the license type. If you currently have a "4 line Enterprise" then you will need to upgrade to "4 line Enterprise+Dialer".



#36 ahmed.muneer 27 October 2017 - 12:16 AM

Dear Vg,

i have successfully registered the IVR as SIP extension on the Avaya IPO but once i do transfer it cause the below error

405 method not allowed.

Please refer to the wireshark and the logs file

 



#37 SupportTeam 27 October 2017 - 08:31 AM

Traces show VoiceGuide registered itself as extension 199.

 

A call was placed on IP Office from extension 380 to extension 199. That call was routed to VoiceGuide and VoiceGuide answered it.

 

About 18 seconds after answering the call the VoiceGuide script was set to transfer the call out using a REFER transfer - but the transfer destination was set as 380@192.168.5.1:5060 - the same extension from which the call was placed into VoiceGuide.

 

Please try transferring the call to a different extension then the one that placed call into VoiceGuide.

 

Also, probably no need to specify ":5060" at end of destination address.

 

 

EDIT: Looking at WireShark trace we can see that the transfer was being made to 112@192.168.5.1

  • ws.png
  • ws_refer.png

Edited by SupportTeam, 29 October 2017 - 09:37 PM.
corrected answer


#38 SupportTeam 27 October 2017 - 08:51 AM

Also ensure that "Call Waiting" option is enabled on the IP Office for all the parties involved.



#39 ahmed.muneer 29 October 2017 - 05:41 PM

Dear Vg Team,

i have already defined different extension to receive the call but i don't know why the IVR keep transferring it to caller ID.

refer to the design and print screen.

  • Call Transfer option.png


#40 SupportTeam 29 October 2017 - 06:30 PM

Please post full traces capturing the service startup, the call and the transfer attempt. We can then advise what the traces are showing.

 

Please include the vgEngine, ktTel and WireShark traces.



#41 ahmed.muneer 29 October 2017 - 06:42 PM

kindly find the requested IVR logs and WS tracing logs



#42 SupportTeam 29 October 2017 - 09:33 PM

ktTel and vgEngine traces show two transfers were made to 198@192.168.5.1.

 

The WireShark trace shows that response from IPOffice was: "405 Method Not Allowed"

 

Is the "Call Waiting" option is enabled on the IP Office for all the parties involved?

 

Are you able to examine the Avaya Logs to see why Avaya did not proceed with the transfer?


830 113008.599  2280   3 fn    TransferBlind (3,0,198@192.168.5.1:5060,87223936)
831 113008.600  2280   3       TelDriver_TransferBlind start
832 113008.600  2280   3       TelDriver_TransferBlind no SIP: or TA: prefix specified. Adding SIP: prefix to the transfer destination.
833 113008.600  2280   3       TelDriver_TransferBlind sXferDestModified=[SIP:198@192.168.5.1:5060]
834 113008.601  2280   3       CallTransfer_Invoke start
835 113008.601  2280   3       gc_InvokeXfer SIP:198@192.168.5.1:5060
836 113008.601  2280   3       gc_InvokeXfer SIP:198@192.168.5.1:5060 ok
837 113008.622  5428   3       ev idx=139 : evttype=873(873)=2163(2163) metaevent.crn=8000001, data=005ABD98(06DE0F60), len=8(8) q: 0/3
838 113008.622  5428   3 ev    GCEV_INVOKE_XFER_REJ (2163 0x873) general handler, raising Event_Dialogic
658 113721.115  2280  10 fn    TransferBlind (10,0,198@192.168.5.1,87223940)
659 113721.115  2280  10       TelDriver_TransferBlind start
660 113721.115  2280  10       TelDriver_TransferBlind no SIP: or TA: prefix specified. Adding SIP: prefix to the transfer destination.
661 113721.115  2280  10       TelDriver_TransferBlind sXferDestModified=[SIP:198@192.168.5.1]
662 113721.115  2280  10       CallTransfer_Invoke start
663 113721.115  2280  10       gc_InvokeXfer SIP:198@192.168.5.1
664 113721.116  2280  10       gc_InvokeXfer SIP:198@192.168.5.1 ok
665 113721.121  5428  10       ev idx=211 : evttype=873(873)=2163(2163) metaevent.crn=8000003, data=041651D0(06DE0F60), len=8(8) q: 0/3
666 113721.121  5428  10 ev    GCEV_INVOKE_XFER_REJ (2163 0x873) general handler, raising Event_Dialogic
  • ws_refer.png


#43 ahmed.muneer 30 October 2017 - 08:28 PM

Dear VG team,

 

i have enabled the Call waiting on all evolved party but still transfer is not working,

Please transfer to the attached logs.



#44 SupportTeam 01 November 2017 - 12:16 PM

Are you able to capture logs on the IPOffice? Maybe those will show why it's responding with a "405 Method Not Allowed"

  • ws.png


#45 ahmed.muneer 02 November 2017 - 04:02 AM

Dears,

kindly find the attached logs from Avaya IPO



#46 SupportTeam 02 November 2017 - 07:02 AM

That trace does not show any information as to why the "405 Method Not Allowed" was sent after receiving the REFER message. There is just tracing of the messages for that interaction and nothing else.

 

Are you able to have Avaya IPO generate logs that contain more tracing information detailing what is happening on that system? We need tracing that shows why Avaya decided to respond with 405 instead of proceeding with transfer.


 20:53:58  193147938mS SIP Rx: UDP 192.168.5.204:5060 -> 192.168.5.1:5060
                    REFER sip:380@192.168.5.1:5060;transport=udp SIP/2.0
                    From: <sip:199@192.168.5.1>;tag=f5a1ff8-0-13c4-65014-440-54db1a29-440
                    To: "Ahmad Alnakhalah - Dev"<sip:380@192.168.5.1>;tag=e9c390943d8310a3
                    Call-ID: ffc82cd9dc2e53174b32896ed3a441ca
                    CSeq: 1 REFER
                    Via: SIP/2.0/UDP 192.168.5.204:5060;branch=z9hG4bK-449-10bdef-60d0577c-f595db8
                    Refer-To: <sip:198@192.168.5.1>
                    Referred-By: <sip:199@192.168.5.204>
                    Max-Forwards: 70
                    Supported: replaces
                    Contact: <sip:199@192.168.5.204>
                    Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY
                    Allow-Events: refer
                    Content-Length: 0
                    
 20:53:58  193147938mS SIP Tx: UDP 192.168.5.1:5060 -> 192.168.5.204:5060
                    SIP/2.0 405 Method Not Allowed
                    Via: SIP/2.0/UDP 192.168.5.204:5060;branch=z9hG4bK-449-10bdef-60d0577c-f595db8
                    From: <sip:199@192.168.5.1>;tag=f5a1ff8-0-13c4-65014-440-54db1a29-440
                    Call-ID: ffc82cd9dc2e53174b32896ed3a441ca
                    CSeq: 1 REFER
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE
                    Supported: timer,100rel
                    Server: IP Office 10.1.0.0.0 build 237
                    To: "Ahmad Alnakhalah - Dev" <sip:380@192.168.5.1>;tag=e9c390943d8310a3
                    Content-Length: 0


#47 ahmed.muneer 02 November 2017 - 04:59 PM

Dear,

i were searching through this and the problem is there some missing in header that from your side the IVR.



#48 SupportTeam 02 November 2017 - 05:45 PM

Can you please post the details of what you are referring to. Do you have a link to documentation or traces of successful REFER transfers etc?