Bart, Did you try the method of inband along with changing the frequencies at the same time?
Thanks, Steve T On Sat, Jun 21, 2008 at 3:29 PM, Barton Fisher <[EMAIL PROTECTED]> wrote: > OK, tried changing DTMF tone as described on URL and no difference > > Bart > > Steve, I fooled with dtmf mode and it was 2833 - However, got stranger > results with inband, seems it would take digits, but audio goes to 1 way > afterwards first push. > > As far as changing the code per the URL, I think I get what's it doing, but > wonder what else would be effected afterwards - I guess I could switch back > if it turns out to be a bad idea > > Bart > > > On Sat, Jun 21, 2008 at 12:11 PM, Barton Fisher <[EMAIL PROTECTED]> wrote: >> I place SIP DID call towards ZAP (TE410P). ZAP uses e&m signaling to an >> external IVR system. I can hear the asterisk sending the DTMFs properly >> toward ZAP at call setup. After the call connects, any further DTMF digits >> from SIP is very short sounding or distorted (barely audible) on the ZAP >> and ignored. ZAP to ZAP connections work perfect. >> >> Just so you know, with 1.2 this is not an issue and this issue is keeping > me >> from moving to 1.4. >> >> I have a test system setup with a SIP DID to a test IVR system to >> demonstrate this problem. I can provide full access to these systems for >> testing. I've placed on Digium bugs but have not received any responses > yet. >> There is nothing in the logs or cli that provides anything meaningful - >> Below is a call where I press '*' and it is ignored. > > Hello, here are a few pointers that might help. Are you using > RFC2833, inband, info? My guess is 2833, you might want to give > inband a try unless you are using a lossy codec. > > This is pretty interesting and might solve your issue. It seems that > by doing this, Asterisk just passes the DTMF as regular audio instead > of trying to interpret it. Bookmarked for when I run into this same > issue..... > http://astrecipes.net/index.php?n=248 > > Linked from this page on the wiki that has more info on your issue. > http://www.voip-info.org/wiki/view/Asterisk+DTMF > > Thanks, > Steve Totaro > > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
