This is not an answer but rather an addition to the question. We're using a large scale VOIP only asterisk system that has PAP2 enduser units using inband as their DTMF mode. sip.conf is set for using inband as well, and we pass PSTN calls through a provider.
Here's the problem, when our users call other IVR systems like banks and other voicemails they have been unable to always pass the keys they press. Sometimes if they press the keys slowly it works, but not all the time, and otherwise it definitely doesn't work. Anyone else had this problem and/or know of a possible solution? Sherwood ->-----Original Message----- ->From: [EMAIL PROTECTED] ->[mailto:[EMAIL PROTECTED] On Behalf Of maka ->Sent: Tuesday, August 16, 2005 8:34 AM ->To: Asterisk Users Mailing List - Non-Commercial Discussion ->Subject: Re: [Asterisk-Users] Issue with DTMF Tones - Codec Issues -> ->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
