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
