No. sipXbridge does not have a way to decode this. Moving to an ITSP who adheres to SIP standards would be a good move.
voip.ms is readily available where you are I believe, and you can port only the number(s) you need and be up in a matter of minutes. You could also have the existing ITSP forward your calls to a newly established number with another ITSP to get around the DTMF while your number ports. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: [email protected] <[email protected]> To: [email protected] <[email protected]> Sent: Tue Sep 07 07:50:28 2010 Subject: [sipx-users] SipXbridge decode of DTMF Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <51573> Message-ID: <[email protected]> My ITSP is very good, but as with all of them, there is one feature which is quite annoying. They send all DTMF inband and don't use RFC2833. As a result, I can't access any AA features when calling in. I have pointed out this deficiency but it has now been six months with no resolution. Is there any way to get sipXbridge (or is it sipXrelay?) to decode audio and generate RFC2833 codes based on the audio? I really don't want to change ITSPs and figure there might be more than one way to skin a cat! Keith _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
