tracinet
2007-Jun-26 14:21 UTC
[asterisk-users] rcf2833 DTMF broken in asterisk SIP channel?
I posted this bug yesterday: http://bugs.digium.com/view.php?id=10058 but really was hoping that one of you would be willing to try something simple for me and reply back with your results. Basically - I have run into a problem where Asterisk RFC2833 DTMF does not seem to be compatible with large SIP providers such as Level 3 and Global Crossing. Can someone who is using rfc2833 DTMF with a non-asterisk SIP provider try inserting this in their dial plan to see if it works (trying to see if the DTMF tones are heard on the PSTN side of the equation). What happens to me is the first digit gets heard but then silence as the next 8 digits are sent. extensions.conf: exten => 5555555555,1,Dial(SIP/sip_provider/5555555555,20,D(123456789)) For your info - here is what I have in my sip.conf: [general] disallow = all allow=ulaw port = 5060 context = incoming maxexpirey=180 defaultexpirey=160 canreinvite=no srvlookup=yes videosupport=no nat=no tos=reliability dtmfmode=rfc2833 [sip_provider] type=friend username=123456789 secret=password host=10.0.0.1 disallow=all allow=ulaw maxexpirey=15 relaxdtmf=yes dtmfmode=rfc2833 nat=no insecure=very canreinvite=no promiscredir=yes Thanks in advance for your help! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070626/1cf27b06/attachment.htm