Also, when you tried inband, did you set it on the phone as well as sip.conf?
Thanks, Steve T On Sun, Jun 22, 2008 at 10:35 AM, Steve Totaro <[EMAIL PROTECTED]> wrote: > 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
