BCM50 5.0 Skype SIP Trunking - Duplicate DTMF on AA?

[dhnl]

Hello,

We have successfully implemented SIP trunkinkg to Skype with a BCM50 R5.0 (latest patches).

I followed Nortel's configruation guide:

http://support.nortel.com/go/main.jsp?cscat=DOCDETAIL&id=1000425&poid=17141

However, we are unable to use the automated attendant for incoming calls from the PSTN. It seems digits entered are "doubled" where the system recognizes "1" as "11" or # as "##"

Dialing out works perfectly fine. If I call from within skype (using their soft phone) it also works perfectly fine.

I've tried forcing the system to G729 and G711 with no success either way.

Any ideas what might be causing this? Skype support has been helpful but they're stumped as well.

 

[dhnl]

Found a solution for this.

Just in case anyone else runs into the same issue:

1. Navigate to Configuration -> Resources -> IP Trunks -> General
2. Click on the "IP Trunks Settings" tab
3. Ignore in-band DTMF in RTP: by clicking on the check box next to it.

 

[JA939384]

Hi, I am very insterested how you got the BCM/Skype integration working. I have been following the guide here -

http://support.avaya.com/css/P8/documents/100114324

- however I seem to be lacking the Skype template for BCM50.

Have you any pointers / guides to get this working with Skype Connect?

Thanks
Jeremy