On Tue, 10 Aug 2004, AJ Grinnell wrote: > I hadnt heard of that setting until today either, but it still doesnt work. > I am using dtmfmode-rfc2833 in sip.conf, and I have my spa's set to avt or > auto. The "internal" dtmf used for voicemail or anything within * works just > fine. I hust cant get tones out to the PSTN. The DTMF sounds very distorted > on the other end of the call. I am trying to use rfc2833. If you have any > ideas, please let me know. Thank you.
it wasn't until a few minutes after I posted that I realized I should have recommended using 'sip show channel ...' while the call is active. Asterisk will tell you which DTMF mode it's trying to use on the channel. That's how I initially discovered the setting for Broadvoice users to receive dtmf on inbound calls.. I knew it should be inband, but sip show channel told me that asterisk was using rfc2833 for that leg of the call. I went looking for a way to change that setting and found a solution to the problem. If your codec supports inband, you could give it a try.. (I prefer out-of-band too, but if the endpoint only supports inband... then you can't beat 'em and end up joining 'em.) oh, you said "dtmfmode-rfc2833" above. That was a typo, right..? You really have a '=' rather than '-' in sip.conf? as for SIPDtmfMode, try a case-insensitive grep on the source to see if that string comes up anywhere. If it isn't there, then it's not a real option and would probably be ignored by the config-parsing algorithm. Greg > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Greg Hill > Sent: Tuesday, August 10, 2004 1:32 PM > To: Asterisk > Subject: Re: [Asterisk-Users] DTMF issues > > > On Tue, 10 Aug 2004, AJ Grinnell wrote: > > > I am now at a total loss. Using Sipura spa-2000s connected to *, I get > > DTMF working just fine for internal extensions, voicemail, etc. If > > making an outgoing call like this spa --> * --> Cisco AS5350 --> PSTN, I > > get no dial tone. I am working unsuccessfully with Cisco right now on > > this, but they cant find anything wrong. I have tried all suggestions I > > can find from the list and elsewhere. I have added SIPDtmfMode to my > > outgoing extensions, that still doesnt help. Does anyone out there have > > experiance or ideas with this setup? > > I haven't heard of any SIPDtmfMode setting, but there is dtmfmode= (in > sip.conf, not extensions.conf). Which dtmf mode are you hoping to use? _______________________________________________ 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
