Support Request: Voip CALL Transfer not working

Description

Hello,
We have a problem with our voip call. We enter all the settings for the SIP settings, we made the call to our Call Center. There we have an IVR. The call gets connected on the IVR and after our agent pick up the call we have one way audio: I can hear them, but they don't. If we place a direct call to the agent (without passing the IVR) the call is fine, so the problem is when we hit the IVR and the agent pick up the call.

We put a trace (which it's attached here) and we find out that the media RTP si send only from the agent because the application does not send the ACK packet.

Please let me know if you need another information.

Thanks,
Adrian

Answer: (3)

Re: Voip CALL Transfer not working 4/16/2015 11:54 AM
Hello,

The VoIP feature of SiteKiosk is implemented with an external SIP SDK (PortSIP SDK 11.1) and only was tested with usual SIP gateways on the internet but not with special IVR hardware.

You may firs check it with using the PortGo Softphone for Windows that can be downloaded from the home page of the manufacturer of the SDK:
http://www.portsip.com/downloads.html

Regards,
Michael Olbrich
Re: Voip CALL Transfer not working 4/16/2015 4:56 PM
Hello,

I did download that soft phone, but the problem is the same. One of my colleagues already contact portsip. com with an email explaining them the issue we have.
Is that possible that you can escalate this email to a higher priority?

Thanks,
Adrian
Re: Voip CALL Transfer not working 4/17/2015 8:46 AM
Hello,

I can’t do this because we just implemented the PortSIP SDK into our SiteKiosk software but we are not the developer of the PortSIP SDK and have no direct influence to the development of this SDK (we are not PortSIP).

Using VoIP via SIP is one of a lot of features in SiteKiosk:
http://www.provisio.com/web/us/products/windows-kiosk-software-sitekiosk

Regards,
Michael Olbrich
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