just a suggestion, but why don't you try using RFC2833 dtmf relay between the cisco and the asterisk box.
use dtmfmode=rfc2833 in sip.conf, and you can also set the dtmf mode per peer in sip.conf also, if you use inband dtmf, this would only work with u-law and a-law, and not g729. on the cisco, enter Router(config-dial-peer)# dtmf-relay rtp-nte in dial-peer configuration mode. I recently had problems with a cisco gw forwarding pstn dtmf digits to my asterisk box, and rfc2833(which is what rtp-nte stands for in cisco's terms) solved it successfully. cheers On 8/16/05, Aaron W <[EMAIL PROTECTED]> wrote: > Topology: > PSTN<-T1 PRI->NEAX2400<-T1 PRI->Cisco 3825<-Ethernet-> Asterisk VoIP server > > When I make a call to a VoIP user from the PSTN, the call gets routed > through the PBX, and Cisco. Because of that the DTMF tones are passed > inband, which I can hear on the VoIP end of the call. However, I have > one extension on asterisk set up so that I can check voice mail when > away from my phone. When I call that number again via the PSTN, and I > am prompted to enter my extension number Asterisk never "hears" the > dtmf tones. I have done some digging around, and my guess is that the > issue relates to the codec being used messing up the tones. > > Am I on the right track? Is there a ideal way to handle this? what do > others do? > > I have posted my sip.conf below. > > Thanks, > Aaron > > [general] > port = 5060 ; Port to bind to > bindaddr = 0.0.0.0 ; Address to bind to > context = default ; Default for incoming calls (default > context has no routing for security purposes) > ;dtmfmode=rfc2833 > dtmfmode=inband > srvlookup = yes > disallow=all ; Disallow all codecs > ;allow=g729 ; Codecs that we allow (in order of preference) > allow=ulaw > ;allow=alaw > allow=g729 > ;allow=ulaw > ;allow=all > > > [3120] > callerid=Aaron Walsh <3120> > type=friend > host=dynamic > canreinvite=no > qualify=yes > nat=yes > setvar=LDPREFIX=1999999 > context=XXXXXXX > secret=XXXXX > [EMAIL PROTECTED] > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- I'm sick and tired of being sick and tired... _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
