Thanks I give give that a try. One follow up question. If the call is coming in via the PSTN, and going through the NEAX (PBX) then to the Cisco, can I control the way the PBX sends the DTMF, or is the cisco some how able to split out the DTMF tones from everything else?
I was assuming that becuase I am going through the PBX, the cisco would recieve the DTMF inband, and therefore it would have to send it out also as inband. Thanks again Aaron On 8/16/05, maka <[EMAIL PROTECTED]> wrote: > 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 > _______________________________________________ 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
