Hi All, Anyone have any idea what could cause incoming calls on a SIP channel to no longer be able to use DTMF? DTMF on incoming calls on zaptel and on local SIP softphones and ATAs all work fine. Nothing gets registered in the CDR or on the console in verbose level 10, it just times out. I haven't changed anything on my part and can't get through to Viatalk tech support to ask them what they changed (fat load of luck getting that question answered anyway). Everything was working fine with dtmfmode=inband and relaxdtmf at the default, now I've tried all 6 valid combos of those two settings with no change. This is on asterisk 1.2.27 that's been working fine in production for about 3 months now.
Here's the section from sip.conf (the way it had been working all along): [viatalk] type=peer secret=(yep it's right) username=(yep it's right) host=newyork-1.vtnoc.net canreinvite=no insecure=very qualify=yes context=incoming-viatalk dtmfmode=inband ; Choices are inband, rfc2833, or info ;relaxdtmf=yes ; Relax dtmf handling Thanks in advance for any help. I've got all incoming calls on Viatalk shunted to an extension in the meantime, not an elegant solution. Best regards, David [EMAIL PROTECTED] ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs _______________________________________________ -- 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