mario staphorst wrote: > Thank you for your reply. > I understand that Asterisk is not a SIP proxy, but shouldnt this header > be passed on in order to provide proper T.38 passthrough support in this > case? > As far as i can see is this header really needed to make the T.38 > connection successfull, when i setup the call directly to the ATA the > reinvite is going fine.
T.38 negotiation has already been improved in later releases than what you are using, so I'd suggest upgrading to 1.4.25 (or 1.4.26-rc1) before continuing, as it is possible that your issue has already been fixed. However, there are still areas we've identified where our T.38 negotiation needs some additional work, and we'll be trying to address those shortly. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: [email protected] Check us out at www.digium.com & www.asterisk.org _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
