Alcatel Unleashed

Hi,

I have a customer who is having trouble with DTMF selection when a call received on a OXE SIP extension ( Polycom 7000) from a webex conference. For them to enter the conference, they need to dial 1 but this does not work.

Outbound calls from this phone DTMF works fine.
For calls to normal IP phones ( 4028/4038) on the OXE, auto DTMF for inbound calls ( System Parameters) is switched TRUE and has been working for inbound Webex calls.

Have asked user to dial ## and then test, it seems like the digits come through after a certain timer but it looks like the call is put on hold and the user being unable to retrieve the call back.

The SIP gateway is configured with payload of 101. Cannot find any paramaters in the Polycom 7000 itself.

Anyone come across DTMF selection on an inbound call to a SIP Extension?

Thanks

OXE is common hardware with R10.0 software.

what about if you decrease dtmf payload from 101 to 97 ?
also in ip 6000 i don't found where to mange dtmf , but in the technical note datasheet of ip 7000 we can read
in the audio caracteritics dtmf generation /rtp flux for dtmf