Call transfers with NV6310

[Anonymous]

How do you perform a call transfer (outside of building) with a NV6310 with FXS lines? I am running the latest version A5.03.00. Our sip server will not provide the features since it is sip lines, so I enabled these on the 6310. I also enabled 3WC on the 6310.

[Anonymous]

Hello and thank you for posting to our forum. 

The call transfer and rtp media mixing will have to be handled by a Sip Server or PBX.  The NV 6300s and TA 900s were designed to work in tandem with the SIP Server to handle an FXS analog user trying to transfer or conference in another user.  With the flash mode set to transparent in the ADTRAN, when an FXS users hits the flashhook, the ADTRAN sends a SIP Info message to the SIP Server with an Event Flashhook in the message.  When the ADTRAN is set to flash mode interpreted, the ADTRAN sends a SIP Invite to conference URI when an FXS user hits the flash button.  Either way, a SIP Server is needed for the functionality you require.

Let me know if you have any more questions.

Regards,

Geoff

[Anonymous]

Thanks for the info. It appears to be working ok for the customer.

[Anonymous]

I am glad it is working.   You may be able to get an unsupported feature to work in some capacity but please note that the feature is not technically supported. 

Also, I wanted to clarify the flash mode interpreted I mentioned earlier, as I did not provide enough details.  What actually happens, is when there is a connected call, the FXS user flashes, putting the original call on hold.  The ADTRAN will send an Invite to the SIP Server for the hold.  There is dial tone after the flash, so the user dials the 2nd party.  After this call connects, the user flashes again.  This time the ADTRAN will send Refers to the SIP Server for both connected calls.  At that point, the SIP Server is responsible for the conference or transfer.

Regards,

Geoff

[Anonymous]

Hi Geo, I am doing some testings with a customer with Huawei IMS and have some problems on the 3 parties conference call. The configuration is such that all conference is handled by the SIP server(network) and is configured as interpreted for the flash hook, do we need to configure anything on the conference URI in the NV6330? The call can be placed to hold when the first flash hook was activated and dial tone was heard and second call can be connected but when the second flash hook was issued, the call cannot be joined to the first call. Appreciate you advice what will be the possible problem. Normal outgoing or incoming call from FXS port of NV6330 can be made without any problem to another phone connected via IMS server, whether is a SIP or a analogue phone

Best regards,

Chan.

[Anonymous]

Hello Chan,

It may be best to get a debug to determine what is happening during this call. Can you get the following debug of the call you are trying to make:

debug sip stack message

debug voice verbose

debug int fxs

Please run all 3 debugs on the same session and either a) increase the session output buffer or b) log the session output to a .txt file.  This way none of the debug will be cut off.  Also, make your session window large enough where the lines do not get truncated.  When you have the file, feel free to send it over.

It may be best to open a ticket with us in Technical Support with this information.  This way your IP addresses and Phone numbers will not be on the forum.  It would be fairly time consuming to change all that information in a large debug. 

Let me know what you would like to do.

Regards,

Geoff

[Anonymous]

Hello Chan,

I went ahead and flagged this post as "Assumed Answered".  If any of the responses on this thread assisted you, please mark them as Correct or Helpful as the case may be with the applicable buttons.  This will make them visible and help other members of the community find solutions more easily.  If you still need assistance, I would be more than happy to continue working with you on this - just let me know in a reply.

Thanks,

Geoff