I would appreciate it if you didn't top-post. das sandesh <sandesh...@gmail.com> writes:
> Hi Benny... > > DTMF tones are heard at the SIP phones side and not the other > party.......'server side' means from the Asterisk side.....from the > wireshark captures it appeards that the dtmf digits were sent from the > asterisk server ip to the phone ip randomly through Cisco(just passes the > SIP packt) inbetween the conversation....... How do you interface with the PSTN? A Digium card? Either way you may want relaxdtmf=no in dahdi.conf if you don't have that already. You can see the DTMF happening on the Asterisk console if you set verbosity high enough. /Benny -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
